From deepaknivas at rediffmail.com  Thu Mar  1 05:06:57 2007
From: deepaknivas at rediffmail.com (Deepak nivas)
Date: 1 Mar 2007 10:06:57 -0000
Subject: [SIPForum-discussion] query
Message-ID: <20070301100657.2816.qmail@webmail99.rediffmail.com>

  Hi,


 What is mean by loose routing?


Regards,
Deepak.
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From williamp at TechMahindra.com  Thu Mar  1 06:18:23 2007
From: williamp at TechMahindra.com (William Prusty)
Date: Thu, 1 Mar 2007 16:48:23 +0530
Subject: [SIPForum-discussion] loose routing
Message-ID: <5C5A863D4858FC41818EF1C777DB6CFA033CD030@SINBNGEX001.TechMahindra.com>


 

 

Strict Routing and Loose Routing

The way how record routing works has evolved. Record routing according
to RFC2543 rewrote the Request-URIi
<http://www.iptel.org/glossary#term81> i
<http://www.iptel.org/glossary#term80> . That means the Request-URI
always
contained URI of the next hop (which can be either next proxy server
which inserted Record-Route header field or destination user agent).
Because of that it was necessary to save the original Request-URI as the
last Route header field. This approach is called strict routing.

Loose routing, as specified in RFC3261, works in a little bit different
way. The Request-URI is no more overwritten, it always contains URI of
the destination user agent. If there are any Route header field in a
message, than the message is sent to the URI from the topmost Route
header field. This is significant change--Request-URI doesn't
necessarily contain URI to which the request will be sent. In fact,
loose routing is very similar to IP source routing.

Because transit from strict routing to loose routing would break
backwards compatibility and older user agents wouldn't work, it is
necessary to make loose routing backwards compatible. The backwards
compatibility unfortunately adds a lot of overhead and is often source
of major problems.

Regards,

william

 



============================================================================================================================

Tech Mahindra, formerly Mahindra-British Telecom.
 
Disclaimer:

This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review at <a href="http://www.techmahindra.com/Disclaimer.html">http://www.techmahindra.com/Disclaimer.html</a> externally and <a href="http://tim.techmahindra.com/Disclaimer.html">http://tim.techmahindra.com/Disclaimer.html</a> internally within Tech Mahindra.

============================================================================================================================
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From NXDR43 at motorola.com  Thu Mar  1 06:51:28 2007
From: NXDR43 at motorola.com (Dawn Somen-NXDR43)
Date: Thu, 1 Mar 2007 19:51:28 +0800
Subject: [SIPForum-discussion] query
Message-ID: <40E89886C8B3B54B98C5291646C591AA014B6206@ZMY16EXM67.ds.mot.com>

HI!
 
Can someone tell me if no expiration is present, is it 3600sec by
default or the server chooses a time interval for that as stated in sec
10.2.1.1 of RFC3261?
Also, is 360 configured locally?
 
Thanks!
Regards,
Somen
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From nirk at MICROSOFT.com  Thu Mar  1 07:58:28 2007
From: nirk at MICROSOFT.com (Nir Katz)
Date: Thu, 1 Mar 2007 12:58:28 +0000
Subject: [SIPForum-discussion] Interoperability Testing - Best course of
	action
Message-ID: <59DD872D2D837D44B60E6B6C630CE4B212D7A8CB68@EA-EXMSG-C303.europe.corp.microsoft.com>

Hi,

Based on your experience what is the best way to test SIP Application Layer Gateway ability to work with other (especially hardware) SIP solutions? Is there any sense in trying to automate the work with different hardware? Or should the focus be on RFC compliance with sporadic and direct manual testing of the various SIP solutions interoperability?

Thanks in advance

Nir Katz


From francesco.landolfo at gmail.com  Thu Mar  1 09:14:23 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Thu, 1 Mar 2007 15:14:23 +0100
Subject: [SIPForum-discussion] About Header Contact
In-Reply-To: <d08f948c0703010608x462c0f39u4f4ba351e143fc37@mail.gmail.com>
References: <d08f948c0703010608x462c0f39u4f4ba351e143fc37@mail.gmail.com>
Message-ID: <d08f948c0703010614g5c962a46s4de2a21fb0bbb1d1@mail.gmail.com>

Hi,
I have the following doubt.
Suppose that you have two client:

   - A (sip:a at mydomain.org);
   - B ( sip:b at mydomain.org).

Suppose you have a Sip Proxy Server too.

I want to setup a call session using Sip Proxy Server. (Please see the
attached file)

When B sends 180 RINGING back to the client A, does it writes in the Contact
header its sip address  (Contact: "B" < sip:B at mydomain.org>) or its address
of care (Contact: "A" < sip:B at 100.101.102.103>) ???

What is the standard method?

Thanks,
Francesco Paolo Landolfo

-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
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From francesco.landolfo at gmail.com  Thu Mar  1 09:08:37 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Thu, 1 Mar 2007 15:08:37 +0100
Subject: [SIPForum-discussion] About Header Contact
Message-ID: <d08f948c0703010608x462c0f39u4f4ba351e143fc37@mail.gmail.com>

Hi,
I have the following doubt.
Suppose that you have two client:

   - A (sip:a at mydomain.org);
   - B ( sip:b at mydomain.org).

Suppose you have a Sip Proxy Server too.

I want to setup a call session using Sip Proxy Server. (Please see the
attached file)

When B sends 180 RINGING back to the client A, does it writes in the Contact
header its sip address  (Contact: "B" < sip:B at mydomain.org>) or its address
of care (Contact: "A" <sip:B at 100.101.102.103>) ???

What is the standard method?

Thanks,
Francesco Paolo Landolfo

-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
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From deepaknivas at rediffmail.com  Thu Mar  1 22:14:30 2007
From: deepaknivas at rediffmail.com (Deepak nivas)
Date: 2 Mar 2007 03:14:30 -0000
Subject: [SIPForum-discussion] query
Message-ID: <20070302031430.13857.qmail@webmail106.rediffmail.com>

  
Hi,


Can any explain conference call flow with a diagram?


Regards,
Deepak.

Regards,
Deepak.
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From zeroroot at tmax.co.kr  Fri Mar  2 03:49:52 2007
From: zeroroot at tmax.co.kr (Young-Geun Park)
Date: Fri, 2 Mar 2007 17:49:52 +0900
Subject: [SIPForum-discussion]  recommanding a tester for sip server
Message-ID: <000001c75ca7$bebeaf10$3d01a8c0@zeroroot>

How about SIPp(sipp.sourceforge.net) for a sip tester?

 

Regards,

Park

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From nicolas.destor at orange-ftgroup.com  Fri Mar  2 09:51:36 2007
From: nicolas.destor at orange-ftgroup.com (zze-DESTOR Nicolas RD-SIRP-LAN)
Date: Fri, 2 Mar 2007 15:51:36 +0100
Subject: [SIPForum-discussion] query
In-Reply-To: <20070302031430.13857.qmail@webmail106.rediffmail.com>
Message-ID: <BAF83494CE653943A97B9F755016A06604A8661C@ftrdmel1.rd.francetelecom.fr>

Hi, 
 
Does someone know a very simple open-source voice-mail server in JAVA or
C++ ( GUI and record functionnality is not nescessary)? 
It's to do somes modifications on it after, but I'm not an expert
programmer so I'm looking for a simple voice-mail before begin coding! 
 
thanks for your help. (I don't ask to you Kaushik, you help me already a
lot!)
 
regards,
Nicolas 
 
 
For information, here the call-flow that the modified voice-mail need to
support. 
 
 
 
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From durgani at gmail.com  Fri Mar  2 10:13:02 2007
From: durgani at gmail.com (Prakash Durgani)
Date: Fri, 2 Mar 2007 10:13:02 -0500
Subject: [SIPForum-discussion] query
In-Reply-To: <BAF83494CE653943A97B9F755016A06604A8661C@ftrdmel1.rd.francetelecom.fr>
References: <20070302031430.13857.qmail@webmail106.rediffmail.com>
	<BAF83494CE653943A97B9F755016A06604A8661C@ftrdmel1.rd.francetelecom.fr>
Message-ID: <f6f514070703020713t5a2b30eawcfa2aeca882a639f@mail.gmail.com>

Have you looked at Asterisk? or maybe a combination of SER (SIP Express
Router) and SEMS (SIP Express Media Server).

On 3/2/07, zze-DESTOR Nicolas RD-SIRP-LAN <nicolas.destor at orange-ftgroup.com>
wrote:
>
>  Hi,
>
> Does someone know a very simple open-source voice-mail server in JAVA or
> C++ ( GUI and record functionnality is not nescessary)?
> It's to do somes modifications on it after, but I'm not an
> expert programmer so I'm looking for a simple
> voice-mail before begin coding!
>
> thanks for your help. (I don't ask to you Kaushik, you help me already a
> lot!)
>
> regards,
> Nicolas
>
>
> For information, here the call-flow that the modified voice-mail need to
> support.
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
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From sivam at motorola.com  Fri Mar  2 10:39:51 2007
From: sivam at motorola.com (Siva M-Q16748)
Date: Fri, 2 Mar 2007 23:39:51 +0800
Subject: [SIPForum-discussion] query
In-Reply-To: <20070301100657.2816.qmail@webmail99.rediffmail.com>
Message-ID: <988EE2C769AC284ABAE9328BFC10703F01815135@ZMY16EXM66.ds.mot.com>

Hi
 
In case of loose routing in the responce sent to a request ,Request-URI
would contain the final destination to be reached and the Route header
is used as the path to reach the same
 
Where as in strict routing the Request-URI would say the next hop to be
reached and the last entry in route will be the final destination to be
reached
 
There are very good examples in RFC3261 Section 16.12
 
Siva

________________________________

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Deepak nivas
Sent: Thursday, March 01, 2007 3:37 PM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] query



  Hi,


What is mean by loose routing?


Regards,
Deepak. 



 
<http://adworks.rediff.com/cgi-bin/AdWorks/sigclick.cgi/www.rediff.com/s
ignature-home.htm/1507191490 at Middle5?PARTNER=3>  
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From nicolas.destor at orange-ftgroup.com  Fri Mar  2 11:12:26 2007
From: nicolas.destor at orange-ftgroup.com (zze-DESTOR Nicolas RD-SIRP-LAN)
Date: Fri, 2 Mar 2007 17:12:26 +0100
Subject: [SIPForum-discussion] query
In-Reply-To: <f6f514070703020713t5a2b30eawcfa2aeca882a639f@mail.gmail.com>
Message-ID: <BAF83494CE653943A97B9F755016A06604A86699@ftrdmel1.rd.francetelecom.fr>

Sorry for my late answer.
Yes I looked at Asterisk but It's not what I want. The voice-mail will be install on a existing SIP network (the sip server is already present). In fact the installation process have to be the same than a sipphone, the alone difference is that the voice-mail answer automaticelly when it receive a INVITE message!  
I don't know SER and SEMS but I think is the same problem...
 
But thanks for your response! If you have anothers ideas says me !
 


________________________________

De : Prakash Durgani [mailto:durgani at gmail.com] 
Envoy? : vendredi 2 mars 2007 16:13
? : zze-DESTOR Nicolas RD-SIRP-LAN
Cc : discussion at sipforum.org
Objet : Re: [SIPForum-discussion] query


Have you looked at Asterisk? or maybe a combination of SER (SIP Express Router) and SEMS (SIP Express Media Server).


On 3/2/07, zze-DESTOR Nicolas RD-SIRP-LAN <nicolas.destor at orange-ftgroup.com> wrote: 

	Hi, 
	 
	Does someone know a very simple open-source voice-mail server in JAVA or C++ ( GUI and record functionnality is not nescessary)? 
	It's to do somes modifications on it after, but I'm not an expert programmer so I'm looking for a simple voice-mail before begin coding! 
	 
	thanks for your help. (I don't ask to you Kaushik, you help me already a lot!)
	 
	regards,
	Nicolas 
	 
	 
	For information, here the call-flow that the modified voice-mail need to support. 
	 
	
	_______________________________________________
	This is the SIP Forum discussion mailing list
	TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
	Post to the list at discussion at sipforum.org
	
	


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From sicu13 at yahoo.ca  Fri Mar  2 18:16:21 2007
From: sicu13 at yahoo.ca (Sicu Babanul)
Date: Fri, 2 Mar 2007 18:16:21 -0500 (EST)
Subject: [SIPForum-discussion] virtual number forward to my cell
Message-ID: <919882.46090.qm@web63103.mail.re1.yahoo.com>

  I am trying to find a virtual number from Romania that I can forward to my mobile, does anyone know where I can get one?
thank you

  
 		
---------------------------------
Be smarter than spam. See how smart SpamGuard is at giving junk email the boot with the All-new Yahoo! Mail  
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From naveed770 at yahoo.com  Sat Mar  3 01:34:48 2007
From: naveed770 at yahoo.com (naveed khan)
Date: Fri, 2 Mar 2007 22:34:48 -0800 (PST)
Subject: [SIPForum-discussion] Authentication for third party registration
	in SIP
Message-ID: <638333.897.qm@web60019.mail.yahoo.com>

hi to all
   
  Can any one of you tell me that how the authentication is done in case of third party registration.
  And I need your help regarding why there is a need for third party registration in sip.
  How a third party (supposed Bob) will come to know that he has to register on the behalf of say "Alice".
  Is there any method suggessted by ietf for the authentication of third party registration.
  And who is to authenticate in this process either third person in my case "Bob" or the "Alice".
   
  Thanks in advance for any kind of help
   
  Regards,
  Naveed Khan
   

 
---------------------------------
Sucker-punch spam with award-winning protection.
 Try the free Yahoo! Mail Beta.
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From ext.lore.matheau at sncf.fr  Mon Mar  5 03:20:02 2007
From: ext.lore.matheau at sncf.fr (EXT / LORE MATHEAU Franck)
Date: Mon, 5 Mar 2007 09:20:02 +0100
Subject: [SIPForum-discussion] RE :  query
In-Reply-To: <20070302031430.13857.qmail@webmail106.rediffmail.com>
Message-ID: <00DC188807DB90449DD9384493C07F787AC150@s72sdeig073.ig.sncf.fr>

Hy all,

For any call flow that yuo need, follow this link :

 

http://www.tech-invite.com/

 

Franck

 

-----Message d'origine-----
De : discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] 
Envoy? : vendredi 2 mars 2007 04:15
? : discussion
Objet : [SIPForum-discussion] query

 

  
Hi,


Can any explain conference call flow with a diagram?


Regards,
Deepak. 

Regards, 
Deepak.

 <http://adworks.rediff.com/cgi-bin/AdWorks/sigclick.cgi/www.rediff.com/signature-home.htm/1507191490 at Middle5?PARTNER=3> 

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From deepaknivas at rediffmail.com  Mon Mar  5 04:25:59 2007
From: deepaknivas at rediffmail.com (Deepak nivas)
Date: 5 Mar 2007 09:25:59 -0000
Subject: [SIPForum-discussion] query
Message-ID: <20070305092559.5005.qmail@webmail90.rediffmail.com>

  hi,

what is use In-Reply-To header field?


regards,
deepak.



Regards,
Deepak.
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From imyousuf at gmail.com  Mon Mar  5 23:38:30 2007
From: imyousuf at gmail.com (Imran M Yousuf)
Date: Tue, 6 Mar 2007 10:38:30 +0600
Subject: [SIPForum-discussion] About SIP Presence
Message-ID: <7bfdc29a0703052038i7f3ab2dp4a446592d3a5a733@mail.gmail.com>

Dear Forum members,

I have a query regarding SIP Presence flow. I will illustrate my question
with an example.

imran at smartitengineering.com REQUEST to SUBSCRIBE to
imyousuf at smartitengineering.com EVENT: presence
PROXY - Synchronize with NOTIFY to - imran at smartitengineering.com
Now my question is how does imyousuf at smartitengineering.com update his
information to the PROXY?
As far as my understanding goes "imyousuf" will PUBLISH his status to the
PROXY; now if the UA of "imyousuf" does not ALLOW PUBLISH, can the PROXY
REQUEST to SUBSCRIBE to imyousuf at smartitengineering.com EVENT: presence?

Thanks in advance,

Imran M Yousuf
Enterpreneur & Lead Developer
Smart IT Engineering
Dhaka, Bangladesh
Email: imran at smartitengineering.com
Mobile: +880-1711402557
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From michel at extricom.com  Tue Mar  6 03:48:05 2007
From: michel at extricom.com (Michel Bensoussan)
Date: Tue, 06 Mar 2007 10:48:05 +0200
Subject: [SIPForum-discussion] REGISTER: Request or response?
Message-ID: <45ED2AC5.4010303@extricom.com>

Hello

Looking on RFC 3261, paragraph 20. Header Fields, Table 3, we can see 
that the Warning header may only appear in responses (r), and that it is 
optional (o) in the REGISTER (REG) method.

It this a contradiction?
REGISTER isn't by definition a Request and not a response?

Can we use Warning header in a REGISTER message?

If I cannot use Warning, is there a way to transmit a proprietary 
parameter in the REGISTER message?

Thanks.

Regards,
Michel.


From yahoosam at gmail.com  Tue Mar  6 09:43:33 2007
From: yahoosam at gmail.com (Sam Ernest Kumar Sam)
Date: Tue, 6 Mar 2007 09:43:33 -0500
Subject: [SIPForum-discussion] Sam Ernest Kumar has Tagged you! :)
Message-ID: <200703061443.l26EhX4r022368@sipforum.org>

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From Vinoth.E at mobax.com  Tue Mar  6 09:52:41 2007
From: Vinoth.E at mobax.com (vinoth)
Date: Tue,  6 Mar 2007 06:52:41 -0800
Subject: [SIPForum-discussion] query
Message-ID: <200703060652.AA107610166@mobax.com>

Hi.,

        If you havent Specified the Expires Header., it will Expires in 1 Hour by Default.
       
        Also you can Sent an Expires Header to some other duration such as 30 SECs.,etc locally in you SIP Message and you can Send it to the Server.


 Regards,
Vinoth Kumar.
Mobax Networks, Coimbatore.


---------- Original Message ----------------------------------
From: "Dawn Somen-NXDR43" <NXDR43 at motorola.com>
Date:  Thu, 1 Mar 2007 19:51:28 +0800

>HI!
> 
>Can someone tell me if no expiration is present, is it 3600sec by
>default or the server chooses a time interval for that as stated in sec
>10.2.1.1 of RFC3261?
>Also, is 360 configured locally?
> 
>Thanks!
>Regards,
>Somen
>
>
>

From Vinoth.E at mobax.com  Tue Mar  6 10:02:39 2007
From: Vinoth.E at mobax.com (vinoth)
Date: Tue,  6 Mar 2007 07:02:39 -0800
Subject: [SIPForum-discussion] About Header Contact
Message-ID: <200703060702.AA107348016@mobax.com>


Hi., 
    
          It can be anyone as you said., or even Both.,

For Eg:

Contact: sip:199.175.2.192:36000;

or

Contact: sip:euclid at parthenon.athens.gr

or

Contact: mailto:euclid at geometry.org

or (even two or More within the Same Message.)

Contact: sip:euclid at parthenon.athens.gr
Contact: mailto:euclid at geometry.org


Regards,
Vinoth Kumar.
Mobax Networks, Coimbatore.





---------- Original Message ----------------------------------
From: "Francesco Paolo Landolfo" <francesco.landolfo at gmail.com>
Date:  Thu, 1 Mar 2007 15:14:23 +0100

>Hi,
>I have the following doubt.
>Suppose that you have two client:
>
>   - A (sip:a at mydomain.org);
>   - B ( sip:b at mydomain.org).
>
>Suppose you have a Sip Proxy Server too.
>
>I want to setup a call session using Sip Proxy Server. (Please see the
>attached file)
>
>When B sends 180 RINGING back to the client A, does it writes in the Contact
>header its sip address  (Contact: "B" < sip:B at mydomain.org>) or its address
>of care (Contact: "A" < sip:B at 100.101.102.103>) ???
>
>What is the standard method?
>
>Thanks,
>Francesco Paolo Landolfo
>
>-- 
>Ci?? che facciamo in vita riecheggia nell'eternit??...(Il Gladiatore)
>"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
>presto." (C'era una volta in America)
>E adesso so cosa devo fare, devo continuare a respirare perch?? domani il
>sole sorger?? e chiss?? la marea cosa potr?? portare. (Cast Away)
>Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
>
>



From juanmoncel at yahoo.es  Wed Mar  7 03:24:20 2007
From: juanmoncel at yahoo.es (Juan Montero Celador)
Date: Wed, 7 Mar 2007 09:24:20 +0100
Subject: [SIPForum-discussion] Help with SIPp
Message-ID: <000d01c76092$03218a90$2802a8c0@int.satec.es>

Hello,

 

I am using SIPp for testing. I use it on Windows XP. I am having problems
for playing RTP: I have seen in the documentation the command "exec
play_pcap_audio" but it doesn't work, maybe because I am using WinPcap
instead of Pcap. Can anybody help me?

 

Regards,

--------------------------------

Juan Montero

 

"Mi infancia ha sido tan larga que nunca acaba de terminar"

 

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From lingzhi at xszhengda.com  Wed Mar  7 05:41:50 2007
From: lingzhi at xszhengda.com (lingzhi)
Date: Wed, 7 Mar 2007 18:41:50 +0800
Subject: [SIPForum-discussion] VPN sip/h323 voip gateway
Message-ID: <200703071841495898879@xszhengda.com>


Dear,

We are selling vpn voip gateway, 2 fxs & 4 fxs, both with voip client built in. 

VPN voip gateway is very useful in voip blocked area, and we already tested and well applied.

Plz feel free to contact if interested.

Looking forward to hearing from you.

Best regards, 
  
Ling
Marketing Director
+86 574 25713039-602
+86 13336888688, 13362487887
MSN: ling_zhi_ at hotmail.com
Yahoo: ling_zhi28
http://www.xszhengda.com
Sino-data Information Technology Ltd.



From wang.ran at byd.com.cn  Wed Mar  7 20:17:35 2007
From: wang.ran at byd.com.cn (wangran)
Date: Thu, 8 Mar 2007 09:17:35 +0800
Subject: [SIPForum-discussion] sip protocol questions.(two agent can't
	connect)
Message-ID: <OFA8A512A7.22C1B4C7-ON48257298.00072D67@byd.com.cn>

Dear all:

 

We have problem in sip calls, the attachment is capture the network packet.
Somebody who is familiar with sip protocol may help me analyse why the
problem came out.

 

The test environment is as this:

One sip client 192.168.1.26

 The other sip client      192.168.1.233

Ondo server(sip server) 192.168.1.13

 

 

Best of Regards,

 

wangran

 

 ***********************************************************************

BYD TECHFAITH??COMPANY??LIMITED(BTC)

 

Address:3/F,M8 West,NO.1 Jiu Xian Qiao Dong Road,

        Chao Yang District,Beijing,China  100016

PostCode:10016

Mobile: +86-13810362150

Tel: +86-10-58291226

Mail: wang.ran at byd.com.cn

 ***********************************************************************

 

 

 



Powered by BYD Security Gateway.

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From rtskarthik at gmail.com  Wed Mar  7 23:45:31 2007
From: rtskarthik at gmail.com (Karthik Arumugam)
Date: Thu, 8 Mar 2007 10:15:31 +0530
Subject: [SIPForum-discussion] Call transfer
Message-ID: <322cbb920703072045t42258398te880827ca0638f9c@mail.gmail.com>

*Hi All*

        *Scenario: User A on the soft phone makes a call to the PSTN
respondent*.*User A is in call with PSTN respondent. Now User A wants to
transfer call to the User C on the soft phone residing at same area as that
of User A. *
**
*Are there any issues pertaining to the above scenario? **Regarding call
charges, call transfer possibilities*

*Regards*
*Karthik.A*
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From adityaakumar at hotmail.com  Thu Mar  8 04:38:42 2007
From: adityaakumar at hotmail.com (aditya kumar)
Date: Thu, 08 Mar 2007 09:38:42 +0000
Subject: [SIPForum-discussion] Unable to establish a call
Message-ID: <BAY109-F99DB9C601642DCA80EFB8CE790@phx.gbl>

Hi,

I have successfully register the X-lite soft phone with Asterisk server,But 
unable to establish a call with other extension,When I am trying to call 
,got  503 response (service unavailable)
The Asterisk verision use :asterisk-1.2.15
Whenever I am calling to other extension,On asterisk server I have found the 
following messages.can u tell me where I was wrong If anybody hae prior 
experience of this scenario.
---------------------------------------------------
" 8 14:09:28 NOTICE[2981]: app_dial.c:1055 dial_exec_full: Unable to create 
channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/192.168.98.47-097dc098' status is 
'CHANUNAVAIL'"
--------------------------------------------------

Looking for advance comment

Thanks
Aditya

_________________________________________________________________
"Airtel Song Catcher. Get your Hello Tunes instantly" 
http://www.airtel.in/songcatcher/SONG_CATCHER.html


From ip.telephony at hotmail.com  Thu Mar  8 09:24:32 2007
From: ip.telephony at hotmail.com (mshari and abdulmalik KSU)
Date: Thu, 08 Mar 2007 17:24:32 +0300
Subject: [SIPForum-discussion] Sip softphone
Message-ID: <BAY18-F392E82AA3025FA13CC5D380790@phx.gbl>

i need softphone (open source)

also
the Features are:
- using SIP protocol
- for Pocket PC (( windows mobile ))
-support Cisco Call manger.

i need it ,

regards
malokey

_________________________________________________________________
Express yourself instantly with MSN Messenger! Download today it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/


From lucentdave at vip.sina.com  Fri Mar  9 04:03:40 2007
From: lucentdave at vip.sina.com (Jacky.Wang)
Date: Fri, 9 Mar 2007 17:03:40 +0800
Subject: [SIPForum-discussion] Sip softphone
References: <BAY18-F392E82AA3025FA13CC5D380790@phx.gbl>
Message-ID: <000d01c7622a$8d59a780$04edeedd@unix3g>

eyeBeam , a good soft sip-based terminal, maybe it is what you need.
you could down it from internet.


----- Original Message ----- 
From: "mshari and abdulmalik KSU" <ip.telephony at hotmail.com>
To: <discussion at sipforum.org>
Sent: Thursday, March 08, 2007 10:24 PM
Subject: [SIPForum-discussion] Sip softphone


>i need softphone (open source)
> 
> also
> the Features are:
> - using SIP protocol
> - for Pocket PC (( windows mobile ))
> -support Cisco Call manger.
> 
> i need it ,
> 
> regards
> malokey
> 
> _________________________________________________________________
> Express yourself instantly with MSN Messenger! Download today it's FREE! 
> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
> 
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>


From gurudattbalaji at gmail.com  Fri Mar  9 08:09:46 2007
From: gurudattbalaji at gmail.com (gurudatt balaji)
Date: Fri, 9 Mar 2007 18:39:46 +0530
Subject: [SIPForum-discussion] Query on IMS
Message-ID: <85ddb520703090509u1b160f65ob3e5a84efd7fe37a@mail.gmail.com>

Hi All,

With referrence to attachment, the scenario is wrt unregistered user.

When user is not registred in s-cscf, how i-cscf will find terminating
s-cscf and send invite to s-cscf.

HSS doesnot know the serving s-cscf because user is not registered.

In actual scenario, i-cscf fwds invite to s-cscf and then to voice mail (if
any)

Any idea......

Reg
Balaji
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From ravishankar.shiroor at wipro.com  Fri Mar  9 09:07:03 2007
From: ravishankar.shiroor at wipro.com (ravishankar.shiroor at wipro.com)
Date: Fri, 9 Mar 2007 19:37:03 +0530
Subject: [SIPForum-discussion] Query on IMS
In-Reply-To: <85ddb520703090509u1b160f65ob3e5a84efd7fe37a@mail.gmail.com>
References: <85ddb520703090509u1b160f65ob3e5a84efd7fe37a@mail.gmail.com>
Message-ID: <532B18E13CF9E64380EF5FDDE265E071C6E588@blr-m2-msg.wipro.com>

if the user has subscribed to services like voicemail/missed call
notifications (any service that is triggered when the user is not
registered),
the service (from an AS) will register on behalf of the user. it will
unregister when the user himself registers. the control details depend
ofcourse on how the service is designed.
 
regards,
ravi.
 
--
 
Ravishankar. G. Shiroor
Wipro Technologies, Bangalore.
 
ravishankar.shiroor at wipro.com
--
 


________________________________

	From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of gurudatt balaji
	Sent: Friday, March 09, 2007 6:40 PM
	To: discussion at sipforum.org
	Subject: [SIPForum-discussion] Query on IMS
	
	
	Hi All,
	 
	With referrence to attachment, the scenario is wrt unregistered
user.
	 
	When user is not registred in s-cscf, how i-cscf will find
terminating s-cscf and send invite to s-cscf.
	 
	HSS doesnot know the serving s-cscf because user is not
registered.
	 
	In actual scenario, i-cscf fwds invite to s-cscf and then to
voice mail (if any)
	 
	Any idea......
	 
	Reg
	Balaji

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From haloha201 at yahoo.com  Sun Mar 11 23:16:07 2007
From: haloha201 at yahoo.com (ha do)
Date: Sun, 11 Mar 2007 20:16:07 -0700 (PDT)
Subject: [SIPForum-discussion] Question on 1customer(Taxi service) want to
	have 10 calls at the same time
Message-ID: <397842.84239.qm@web32410.mail.mud.yahoo.com>

Hi 

i have trouble on my customer. The customer want to have 10 calls at the same time
i have Siemens product : HiQ 4200, HiQ 8000

Please give me some advise
Thanks

 
---------------------------------
 Get your own web address.
 Have a HUGE year through Yahoo! Small Business.
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From haloha201 at yahoo.com  Mon Mar 12 00:16:14 2007
From: haloha201 at yahoo.com (ha do)
Date: Sun, 11 Mar 2007 21:16:14 -0700 (PDT)
Subject: [SIPForum-discussion] question on recieving 10 calls at the same
	time on 1 phone number
Message-ID: <468435.97372.qm@web32411.mail.mud.yahoo.com>


Hi 

i have trouble on my customer. The customer want to have 10 calls at the same time on 1 phone number
i have Siemens product : HiQ 4200, HiQ 8000

Please give me some advise
Thanks
  

 
---------------------------------
Now that's room service! Choose from over 150,000 hotels 
in 45,000 destinations on Yahoo! Travel to find your fit.
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From hemantmehar at gmail.com  Mon Mar 12 01:56:15 2007
From: hemantmehar at gmail.com (hemant mehar)
Date: Mon, 12 Mar 2007 11:26:15 +0530
Subject: [SIPForum-discussion] what's the procedure for xlite registration???
Message-ID: <a26ae8fb0703112256p182de41i3c85931f1c947d37@mail.gmail.com>

Hi All,

Can anybody please tell me, what is the procedure to register xlite.

please write in detail???

thanks in advance
Hemant
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From zeroroot at tmax.co.kr  Mon Mar 12 05:12:16 2007
From: zeroroot at tmax.co.kr (Young-Geun Park)
Date: Mon, 12 Mar 2007 18:12:16 +0900
Subject: [SIPForum-discussion] according a performance tester for sipservlet
Message-ID: <200703120910.l2C9AsOu023986@sipforum.org>

Hi, all 

 

Is there any performance tester for sipservlet(JSR116)?

 

Regards, 

Park

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From gurudattbalaji at gmail.com  Mon Mar 12 08:23:43 2007
From: gurudattbalaji at gmail.com (gurudatt balaji)
Date: Mon, 12 Mar 2007 17:53:43 +0530
Subject: [SIPForum-discussion] Video Mail
Message-ID: <85ddb520703120523j6ed31d7eu4490ee1f3c07e4e1@mail.gmail.com>

Hi all,

Is there any standard/spec/RFCs for Video Mail.

Trying to integrate Video Mail with IMS.

Pls confirm

Reg
Balaji
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From chuwfan at gmail.com  Tue Mar 13 03:43:29 2007
From: chuwfan at gmail.com (Chuw Fan Lee)
Date: Tue, 13 Mar 2007 15:43:29 +0800
Subject: [SIPForum-discussion] [HELP]about UA to test SIPProxy
Message-ID: <8a5302dd0703130043u9a49f0bxe8e8a4678a2adc7d@mail.gmail.com>

i just have a sip proxy server here, (mjsip),
i try to connect with window messenger 5.1 with remote client,
it connected but the message can't forward to the other client from remote
client.
anyone can help me please?
urgent.
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From Shanmukharao.Makkapati at airtel.in  Tue Mar 13 07:21:51 2007
From: Shanmukharao.Makkapati at airtel.in (Shanmukharao.Makkapati at airtel.in)
Date: Tue, 13 Mar 2007 16:51:51 +0530
Subject: [SIPForum-discussion] Instant Messaging
Message-ID: <OFBACB3D6F.A6CE6106-ON6525729D.003E3E7B-6525729D.003DFBBF@bharti.com>


HI,
   Can any one help regarding the procedure involved in end to end instant 
messaging. Im just new to SIP. I want to know it in brief. can i get help 
from the forum.


This e-mail and any files transmitted with it are for the sole use of the intended 
recipient(s) and may contain confidential and privileged information. If you are 
not the intended recipient, please contact the sender by reply e-mail and destroy 
all copies and the original message. Any unauthorized review, use, disclosure, 
dissemination, forwarding, printing or copying of this email or any action taken 
in reliance on this e-mail is strictly prohibited and may be unlawful.The recipient 
acknowledges that Bharti Airtel Limited or its subsidiaries and associated companies
(collectively "Bharti Airtel Limited"), 
are unable to exercise control or ensure or guarantee the integrity of/overthe contents 
of the information contained in e-mail transmissions and further acknowledges that any views 
expressed in this message are those of the individual sender and no binding nature of the message shall be implied or assumed 
unless the sender does so expressly with due authority of BHARTI AIRTEL LTD.. Before opening any attachments 
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From rjsparks at nostrum.com  Tue Mar 13 11:05:27 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Tue, 13 Mar 2007 10:05:27 -0500
Subject: [SIPForum-discussion] SIPit 20 registration deadline is March 30
Message-ID: <EC28212F-39FF-49D8-BBD4-26C62F737623@nostrum.com>

If you plan to attend SIPit 20 in Antwerp, Belgium April 16-20
and have not yet registered, now's the time. Registration closes in
just over two weeks (March 30).

See www.sipit.net for more information and the registration link.

RjS

From Shanmukharao.Makkapati at airtel.in  Wed Mar 14 00:42:46 2007
From: Shanmukharao.Makkapati at airtel.in (Shanmukharao.Makkapati at airtel.in)
Date: Wed, 14 Mar 2007 10:12:46 +0530
Subject: [SIPForum-discussion] Modules involved in IM
Message-ID: <OFAC42D951.AACF88E2-ON6525729E.0019A397-6525729E.001971C9@bharti.com>


Hi,
  Can i get info about what the modules included in SIP IM project on C & 
Linux platform and what the structure of sdlc inlcuded, how it will be...?
please help me regard to this.



This e-mail and any files transmitted with it are for the sole use of the intended 
recipient(s) and may contain confidential and privileged information. If you are 
not the intended recipient, please contact the sender by reply e-mail and destroy 
all copies and the original message. Any unauthorized review, use, disclosure, 
dissemination, forwarding, printing or copying of this email or any action taken 
in reliance on this e-mail is strictly prohibited and may be unlawful.The recipient 
acknowledges that Bharti Airtel Limited or its subsidiaries and associated companies
(collectively "Bharti Airtel Limited"), 
are unable to exercise control or ensure or guarantee the integrity of/overthe contents 
of the information contained in e-mail transmissions and further acknowledges that any views 
expressed in this message are those of the individual sender and no binding nature of the message shall be implied or assumed 
unless the sender does so expressly with due authority of BHARTI AIRTEL LTD.. Before opening any attachments 
please check them for viruses and defects 
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From deepaknivas at rediffmail.com  Thu Mar 15 03:16:21 2007
From: deepaknivas at rediffmail.com (Deepak nivas)
Date: 15 Mar 2007 07:16:21 -0000
Subject: [SIPForum-discussion] query
Message-ID: <20070315071621.31672.qmail@webmail103.rediffmail.com>

  
Hi,


  In RFC 3261 chapter 17.1.2 deals with Non-INVITE Client Transaction.
for non-invite method proxy or no one will send provisional response.
what is need of Proceeding state. how come it will come from Trying state to Proceeding state if there is no provisonal response.


Thanks in advance.



Regards,
Deepak.
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From arb3 at alcatel-lucent.com  Thu Mar 15 11:56:37 2007
From: arb3 at alcatel-lucent.com (Bajracharya, Amar R (Amar))
Date: Thu, 15 Mar 2007 10:56:37 -0500
Subject: [SIPForum-discussion] SIP-I to SIP-I
In-Reply-To: <20070315071621.31672.qmail@webmail103.rediffmail.com>
References: <20070315071621.31672.qmail@webmail103.rediffmail.com>
Message-ID: <5C5FD7DED22D4D44AA7B9C853B70E06D9EC032@ILEXC1U02.ndc.lucent.com>

I have not seen any RFCs which talk about SIP-I(with ISUP encapsulation)
to SIP-I scenario. RFC3398 and ANSI T1.679 covers mostly on ISUP to
SIP-I and SIP-I to ISUP.

 

Does any body know how to handle the following scenario?

 

If you are acting as proxy and you get 183 with SDP and 183 with ACM
separately in 2 different messages, are you supposed to

 

-          Pass it along to as it is, OR

-          Drop 183 with SDP(as it doesn't contain any ISUP
encapsulation) , OR

-          Combine 2 and send 183 with SDP+ACM

 

Really appreciate it if anybody has any solution for this.

 

Thanks,

Amar Bajracharya

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From deepanshu at huawei.com  Fri Mar 16 03:26:42 2007
From: deepanshu at huawei.com (Deepanshu)
Date: Fri, 16 Mar 2007 15:26:42 +0800
Subject: [SIPForum-discussion] About ENUM
Message-ID: <014101c7679c$72251360$8178a40a@china.huawei.com>

Hi

ENUM query maps a Tel:URI (008613585147627) to a SIP-URI (sip:user at example.com). Can ENUM query work in the reverse direction mapping a SIP:URI to Tel:URI?

In other words i have a SIP-URI i like to know its corresponding Tel:URI, can i use ENUM query for the same.

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC
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From deepanshu at huawei.com  Fri Mar 16 04:40:33 2007
From: deepanshu at huawei.com (Deepanshu)
Date: Fri, 16 Mar 2007 16:40:33 +0800
Subject: [SIPForum-discussion] About ENUM
References: <E0C5A9C102C96D4989B9B4380A2AB1D002D6B859@ecnbjmw511.eapac.ericsson.se>
Message-ID: <017901c767a6$c30e12e0$8178a40a@china.huawei.com>

It is something related with SMS-IM interworking.

When an IM user sends a SIP MESSAGE (IMS domain) to a SMS-only user (CS
domain) then their is a need to find the Tel:URI associated with the SIP-URI
(included in the SIP MESSAGE) of the sender. The Tel:URI is needed to be
incorporated in the SMS message send towards receiver using which the
receiver can respond.

HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC


----- Original Message -----
From: "Rick Yang (GZ/CBC)" <rick.yang at ericsson.com>
To: "Deepanshu" <deepanshu at huawei.com>
Sent: Friday, March 16, 2007 4:24 PM
Subject: RE: [SIPForum-discussion] About ENUM


Hi,
May I ask what's the driven force behind this idea?


-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Deepanshu
Sent: 2007??3??16?? 15:27
To: discussion at sipforum.org
Subject: [SIPForum-discussion] About ENUM

Hi

ENUM query maps a Tel:URI (008613585147627) to a SIP-URI
(sip:user at example.com). Can ENUM query work in the reverse direction mapping
a SIP:URI to Tel:URI?

In other words i have a SIP-URI i like to know its corresponding Tel:URI,
can i use ENUM query for the same.

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC



From deepanshu at huawei.com  Fri Mar 16 05:19:13 2007
From: deepanshu at huawei.com (Deepanshu)
Date: Fri, 16 Mar 2007 17:19:13 +0800
Subject: [SIPForum-discussion] About ENUM
References: <E0C5A9C102C96D4989B9B4380A2AB1D002D6B8FA@ecnbjmw511.eapac.ericsson.se>
Message-ID: <01b201c767ac$2a17b0e0$8178a40a@china.huawei.com>

Okey,

Thanks for you comments

But, one more issue. It is possible in IM that a user is using a PC or PDA
instead of mobile. In this case i believe he must not have a Tel:URI.
right??

So, how to do in this condition? any ideas

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC

----- Original Message -----
From: "Rick Yang (GZ/CBC)" <rick.yang at ericsson.com>
To: "Deepanshu" <deepanshu at huawei.com>
Sent: Friday, March 16, 2007 4:59 PM
Subject: RE: [SIPForum-discussion] About ENUM


I c.
During the registration, the SIP URI of sender would be registered as well
as the tel URL in the S-CSCF (Implicit registration).
So the tel URL is easy to get from the S-CSCF when it's necessary (IM case).

No ENUM should be concerned I belive.


-----Original Message-----
From: Deepanshu [mailto:deepanshu at huawei.com]
Sent: 2007??3??16?? 16:41
To: Rick Yang (GZ/CBC)
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] About ENUM

It is something related with SMS-IM interworking.

When an IM user sends a SIP MESSAGE (IMS domain) to a SMS-only user (CS
domain) then their is a need to find the Tel:URI associated with the SIP-URI
(included in the SIP MESSAGE) of the sender. The Tel:URI is needed to be
incorporated in the SMS message send towards receiver using which the
receiver can respond.

HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC


----- Original Message -----
From: "Rick Yang (GZ/CBC)" <rick.yang at ericsson.com>
To: "Deepanshu" <deepanshu at huawei.com>
Sent: Friday, March 16, 2007 4:24 PM
Subject: RE: [SIPForum-discussion] About ENUM


Hi,
May I ask what's the driven force behind this idea?


-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Deepanshu
Sent: 2007??3??16?? 15:27
To: discussion at sipforum.org
Subject: [SIPForum-discussion] About ENUM

Hi

ENUM query maps a Tel:URI (008613585147627) to a SIP-URI
(sip:user at example.com). Can ENUM query work in the reverse direction mapping
a SIP:URI to Tel:URI?

In other words i have a SIP-URI i like to know its corresponding Tel:URI,
can i use ENUM query for the same.

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC





From francesco.landolfo at gmail.com  Fri Mar 16 06:29:57 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Fri, 16 Mar 2007 11:29:57 +0100
Subject: [SIPForum-discussion] SIP - Instant Messaging with Session
Message-ID: <d08f948c0703160329r48298459ma99634992d49d2ab@mail.gmail.com>

Hi,
I have read about the possibility to make IM with SIP session.
I'd like to have some other information about this solution and in
particular about a Many to Many chat with Session.

Please can someone give me some links, papers, RFCs... in which I can learn
this things?

If someone does not understand what I mean, please contact me.

Thanks in avoidance,
Francesco Paolo Landolfo

-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
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From erolturac at gmail.com  Fri Mar 16 09:47:07 2007
From: erolturac at gmail.com (erol turac)
Date: Fri, 16 Mar 2007 15:47:07 +0200
Subject: [SIPForum-discussion] SIP - Instant Messaging with Session
In-Reply-To: <d08f948c0703160329r48298459ma99634992d49d2ab@mail.gmail.com>
References: <d08f948c0703160329r48298459ma99634992d49d2ab@mail.gmail.com>
Message-ID: <3079c79a0703160647v6e01a361u6df6b4c9722f647d@mail.gmail.com>

IM is an extension for sip and rfc3428 describes how it should be
implemented.


On 3/16/07, Francesco Paolo Landolfo <francesco.landolfo at gmail.com> wrote:
>
> Hi,
> I have read about the possibility to make IM with SIP session.
> I'd like to have some other information about this solution and in
> particular about a Many to Many chat with Session.
>
> Please can someone give me some links, papers, RFCs... in which I can
> learn this things?
>
> If someone does not understand what I mean, please contact me.
>
> Thanks in avoidance,
> Francesco Paolo Landolfo
>
> --
> Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
> "Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
> presto." (C'era una volta in America)
> E adesso so cosa devo fare, devo continuare a respirare perch? domani il
> sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
> Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>


-- 
Erol Tura?
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From francesco.landolfo at gmail.com  Fri Mar 16 09:55:04 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Fri, 16 Mar 2007 14:55:04 +0100
Subject: [SIPForum-discussion] SIP - Instant Messaging with Session
In-Reply-To: <3079c79a0703160647v6e01a361u6df6b4c9722f647d@mail.gmail.com>
References: <d08f948c0703160329r48298459ma99634992d49d2ab@mail.gmail.com>
	<3079c79a0703160647v6e01a361u6df6b4c9722f647d@mail.gmail.com>
Message-ID: <d08f948c0703160655x33450351q22a4d7a63ea8a0a8@mail.gmail.com>

Yes, I know what you say but RFC 3428 does not describe in details the IM
chat using Session that is what I try.

Thanks in anticipation,
Francesco


Il 16/03/07, erol turac <erolturac at gmail.com> ha scritto:
>
> IM is an extension for sip and rfc3428 describes how it should be
> implemented.
>
>
> On 3/16/07, Francesco Paolo Landolfo < francesco.landolfo at gmail.com>
> wrote:
>
> > Hi,
> > I have read about the possibility to make IM with SIP session.
> > I'd like to have some other information about this solution and in
> > particular about a Many to Many chat with Session.
> >
> > Please can someone give me some links, papers, RFCs... in which I can
> > learn this things?
> >
> > If someone does not understand what I mean, please contact me.
> >
> > Thanks in avoidance,
> > Francesco Paolo Landolfo
> >
> > --
> > Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
> > "Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
> > presto." (C'era una volta in America)
> > E adesso so cosa devo fare, devo continuare a respirare perch? domani il
> > sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
> > Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
> > _______________________________________________
> > This is the SIP Forum discussion mailing list
> > TO UNSUBSCRIBE, or edit your delivery options, please visit
> > http://sipforum.org/mailman/listinfo/discussion
> > Post to the list at discussion at sipforum.org
> >
> >
>
>
> --
> Erol Tura?




-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
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From pavan.jain at ingenient.com  Fri Mar 16 09:58:04 2007
From: pavan.jain at ingenient.com (Pavan Jain)
Date: Fri, 16 Mar 2007 09:58:04 -0400
Subject: [SIPForum-discussion] List of registered user on SIP server
Message-ID: <EXCHANGEunLOyDc41W200000497@mail.ingenient.com>

Hello All:

 

How can I get the list of registered users on the SIP server?

 

Regards,

Pavan

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From wellya at wellya.net  Sat Mar 17 02:24:45 2007
From: wellya at wellya.net (wellya)
Date: Sat, 17 Mar 2007 14:24:45 +0800
Subject: [SIPForum-discussion] List of registered user on SIP server
References: <EXCHANGEunLOyDc41W200000497@mail.ingenient.com>
Message-ID: <200703171424417035768@wellya.net>

you can query database




wellya
2007-03-17



???????? Pavan Jain
?????????? 2007-03-16 22:20:43
???????? discussion at sipforum.org
?????? 
?????? [SIPForum-discussion] List of registered user on SIP server

Hello All:
 
How can I get the list of registered users on the SIP server?
 
Regards,
Pavan
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From wellya at wellya.net  Sat Mar 17 02:26:19 2007
From: wellya at wellya.net (wellya)
Date: Sat, 17 Mar 2007 14:26:19 +0800
Subject: [SIPForum-discussion] SIP - Instant Messaging with Session
References: <d08f948c0703160329r48298459ma99634992d49d2ab@mail.gmail.com>
	<3079c79a0703160647v6e01a361u6df6b4c9722f647d@mail.gmail.com>
	<d08f948c0703160655x33450351q22a4d7a63ea8a0a8@mail.gmail.com>
Message-ID: <200703171426185628520@wellya.net>

Maybe you can search it in http://www.tech-invite.com/ or http://www.rfc-editor.org and http://www.wellya.net 





wellya
2007-03-17



???????? Francesco Paolo Landolfo
?????????? 2007-03-16 22:17:37
???????? erol turac
?????? discussion at sipforum.org
?????? Re: [SIPForum-discussion] SIP - Instant Messaging with Session

Yes, I know what you say but RFC 3428 does not describe in details the IM chat using Session that is what I try.

Thanks in anticipation,
Francesco



Il 16/03/07, erol turac <erolturac at gmail.com> ha scritto:
IM is an extension for sip and rfc3428 describes how it should be implemented. 



On 3/16/07, Francesco Paolo Landolfo < francesco.landolfo at gmail.com> wrote:
Hi,
I have read about the possibility to make IM with SIP session.
I'd like to have some other information about this solution and in particular about a Many to Many chat with Session.

Please can someone give me some links, papers, RFCs... in which I can learn this things? 

If someone does not understand what I mean, please contact me.

Thanks in avoidance,
Francesco Paolo Landolfo

-- 
Ci?? che facciamo in vita riecheggia nell'eternit??...(Il Gladiatore) 
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch?? domani il sole sorger?? e chiss?? la marea cosa potr?? portare. (Cast Away) 
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso) 

_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org





-- 
Erol Tura? 



-- 
Ci?? che facciamo in vita riecheggia nell'eternit??...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto presto." (C'era una volta in America) 
E adesso so cosa devo fare, devo continuare a respirare perch?? domani il sole sorger?? e chiss?? la marea cosa potr?? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso) 
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From danishzaidi54 at yahoo.com  Sat Mar 17 12:33:18 2007
From: danishzaidi54 at yahoo.com (Danish Zaidi)
Date: Sat, 17 Mar 2007 09:33:18 -0700 (PDT)
Subject: [SIPForum-discussion] can any 1 give me samples for CallHold for
	Jain-Sip API
In-Reply-To: <200703171424417035768@wellya.net>
Message-ID: <265790.55020.qm@web90611.mail.mud.yahoo.com>

Can someone give me sample codes for CallHold in JainSip API..

thanx in advance

 
---------------------------------
Food fight? Enjoy some healthy debate
in the Yahoo! Answers Food & Drink Q&A.
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From carol_chang8009 at yahoo.com.cn  Mon Mar 19 02:14:06 2007
From: carol_chang8009 at yahoo.com.cn (Carol Chang)
Date: Mon, 19 Mar 2007 14:14:06 +0800 (CST)
Subject: [SIPForum-discussion] YUXIN news about new model launched
In-Reply-To: <200703171426185628520@wellya.net>
Message-ID: <93030.86408.qm@web15804.mail.cnb.yahoo.com>

   
   YUXIN NEW MODEL YWH201,YWH202 PHONES LAUNCHED
        Through development and experiment for about one year,Zhengzhou Yuneng Communication Co.,Ltd launched a new model YWH202 IP phone,which base on Infineon chip and support SIP,IAX2,H323 protocols,build-in-Router,3-way conference calls,voice message,recording,two SIP servers running at the same time.The highly voice effect ,the friendly menu operation, the easy setting page,and the low fees which can make you enjoy high tech product and save phone expense without too much specialized knowledge.
  At the same time, YWH201 IP Phone based on AR1688 chip and supporting SIP,IAX2 protocol has seized hold of the market by sale of several ten thousands units since it had been released on Oct.2006.This model of phone was developed together by outstanding workers from Aredfox and Zhengzhou Yuneng ,the former is famous in suppling solution in international VOIP field,and the latter has six years excellent performance in sale of internet phones.The type of YWH201 not only inherits merites of the phones based on PA1688,but also has many new functions(side sound,voice message and so on).
  As a researcher and producer in domestic VOIP terminal device,Zhengzhou Yueng communication Co.,Ltd has complete production lines and quality control systems,import and export license,CE and FCC certificate, and also devotes to new technical development and application.It watches colsely the development of VOIP market and the demand of customers ,meanwhile release new products and new functions to make customers satisfied.At present we have YWH100,YWH10,YWH200,YWH300,YWH500(POE)series products based on PA1688 chip,and our YWH200C,YWH300C,YWH600A,YWH600B(True FXO port) support 3-way conference calls,build-in-Router,registering on 3 SIP servers at the same time(switching freely).In future we will release more and more phones which based on AR1688 and Infineon chips.Regarding to gateways,we have had YGW20 based onPA1688,and YGW30A,YGW30B(true FXO port) based on CM5000 chip. These products has been validated and accepted by market.We  cooperate with many companies engaged in
 virtural net and soft switch and other related companies to carry on encryption to solve problems of forcing out from some companies.
  the USB phones(YUS10 and YUS20) and the USB conference box(YHY10) supporting SKYPE and based on C-Media were well appraised by customers and got good sales achievementes as soon as those products were released.
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  At present the ATA YGW50(1FXS0),YGW60(2FXS) which based on Infineon chip has passed testing already and will be put into market soonly.
  We will try our best to provide high quality products and multi-position services for VOIP friends,and make YUXIN first-class brand in the domestic VOIP terminal device! 
   
  -------------------------------------------
ZHENNGZHOU YUNENG COMMUNICATION CO.,LTD
1KANGLELI,ZHONGYUAN DISTRICT,ZHENGZHOU,PRC
URL:HTTP://WWW.YNTX.COM
TEL:86-371-67657240
FAX:86-371-67657239
MSN:Carol_chang8009 at hotmail.com

   
   
   
   
   
   
   


 		
---------------------------------
????????????????-3.5G??????20M?????? 
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From mweiglh at ist.tugraz.at  Mon Mar 19 09:01:25 2007
From: mweiglh at ist.tugraz.at (Martin Weiglhofer)
Date: Mon, 19 Mar 2007 14:01:25 +0100
Subject: [SIPForum-discussion] Proxy-to-User Authentication
Message-ID: <45FE89A5.5060704@ist.tugraz.at>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Dear all,

assume, that a user tries to register a contact address on a registrar
which requires authentication, but the user is not known by the
registrar. The user sends a REGISTER message (without authentication
credentials) which is answered with a "401 Unauthorized" response. Now
the user sends a second REGISTER request which includes an Authorization
header field. What should be the response of the server to the second
REGISTER request? Should the registrar again reply with a "401
Unauthorized", or should the registrar reject the request with a "404
Not Found"?

Thanks in advance.

Regards
Martin

-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.6 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFF/oml6XHVH58yroMRAuH7AKCHsGhkTlLznZKVpLw93FXAqnKIuwCgwCMy
E06PmQIXSxEV62ajCjmDqEk=
=UvKG
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From mweiglh at ist.tugraz.at  Mon Mar 19 09:01:37 2007
From: mweiglh at ist.tugraz.at (Martin Weiglhofer)
Date: Mon, 19 Mar 2007 14:01:37 +0100
Subject: [SIPForum-discussion] Proxy-to-User Authentication
Message-ID: <45FE89B1.8020103@ist.tugraz.at>

Dear all,

assume, that a user tries to register a contact address on a registrar
which requires authentication, but the user is not known by the
registrar. The user sends a REGISTER message (without authentication
credentials) which is answered with a "401 Unauthorized" response. Now
the user sends a second REGISTER request which includes an Authorization
header field. What should be the response of the server to the second
REGISTER request? Should the registrar again reply with a "401
Unauthorized", or should the registrar reject the request with a "404
Not Found"?

Thanks in advance.

Regards
Martin


From rjsparks at nostrum.com  Mon Mar 19 09:35:04 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Mon, 19 Mar 2007 14:35:04 +0100
Subject: [SIPForum-discussion] Proxy-to-User Authentication
In-Reply-To: <45FE89A5.5060704@ist.tugraz.at>
References: <45FE89A5.5060704@ist.tugraz.at>
Message-ID: <7D2E14F7-096F-4246-97B0-392A0E6EEC25@nostrum.com>

It would be an odd edge condition for it to be appropriate for you to  
return
a 404 (given that you returned a 401 the first time). That means that  
whoever
resubmitted the request with credentials has credentials that you are  
willing
to accept as valid for a resource you don't know about. If you have a  
policy
that anyone with an account can modify the registration for any AoR  
on your
system, I could see this happening. Typical systems bind a set of  
credentials
fairly tightly to an AoR (this username password can only be used  
with this AoR
and its the only username password pair I'll accept for that AoR). In  
that situation,
returning a 404 would only make sense if you had recently deleted the  
resource
but hadn't invalidated the credentials yet.

If the credentials in the second request aren't good, you'll return  
another 401.

RjS

On Mar 19, 2007, at 2:01 PM, Martin Weiglhofer wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Dear all,
>
> assume, that a user tries to register a contact address on a registrar
> which requires authentication, but the user is not known by the
> registrar. The user sends a REGISTER message (without authentication
> credentials) which is answered with a "401 Unauthorized" response. Now
> the user sends a second REGISTER request which includes an  
> Authorization
> header field. What should be the response of the server to the second
> REGISTER request? Should the registrar again reply with a "401
> Unauthorized", or should the registrar reject the request with a "404
> Not Found"?
>
> Thanks in advance.
>
> Regards
> Martin
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.6 (GNU/Linux)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFF/oml6XHVH58yroMRAuH7AKCHsGhkTlLznZKVpLw93FXAqnKIuwCgwCMy
> E06PmQIXSxEV62ajCjmDqEk=
> =UvKG
> -----END PGP SIGNATURE-----
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit http:// 
> sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org


From pavan.jain at ingenient.com  Mon Mar 19 10:23:56 2007
From: pavan.jain at ingenient.com (Pavan Jain)
Date: Mon, 19 Mar 2007 10:23:56 -0400
Subject: [SIPForum-discussion] List of registered user on SIP server
In-Reply-To: <200703171424417035768@wellya.net>
Message-ID: <EXCHANGEtpIuPHDXE2s00000589@mail.ingenient.com>

Is there a message format that I need to send to Server( similar to REGISTER)? An example will be helpful.

 

Regards,

Pavan

 

  _____  

From: wellya [mailto:wellya at wellya.net] 
Sent: Saturday, March 17, 2007 2:25 AM
To: Pavan Jain; discussion at sipforum.org
Subject: Re: [SIPForum-discussion] List of registered user on SIP server

 

you can query database

 

  _____  

wellya

2007-03-17

  _____  

???? Pavan Jain

????? 2007-03-16 22:20:43

???? discussion at sipforum.org

??? 

??? [SIPForum-discussion] List of registered user on SIP server

 

Hello All:

 

How can I get the list of registered users on the SIP server?

 

Regards,

Pavan

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From mweiglh at ist.tugraz.at  Mon Mar 19 10:52:22 2007
From: mweiglh at ist.tugraz.at (Martin Weiglhofer)
Date: Mon, 19 Mar 2007 15:52:22 +0100
Subject: [SIPForum-discussion] Proxy-to-User Authentication
In-Reply-To: <7D2E14F7-096F-4246-97B0-392A0E6EEC25@nostrum.com>
References: <45FE89A5.5060704@ist.tugraz.at>
	<7D2E14F7-096F-4246-97B0-392A0E6EEC25@nostrum.com>
Message-ID: <45FEA3A6.1030600@ist.tugraz.at>

Thanks for the fast answer.

If I understood your explanation correctly, that means that I might
mislead the user, such that the user thinks password or username for the
authentication are incorrect. Instead the user might have configured the
wrong registration server on the SIP phone.


Robert Sparks wrote:
> It would be an odd edge condition for it to be appropriate for you to
> return
> a 404 (given that you returned a 401 the first time). That means that
> whoever
> resubmitted the request with credentials has credentials that you are
> willing
> to accept as valid for a resource you don't know about. If you have a
> policy
> that anyone with an account can modify the registration for any AoR on your
> system, I could see this happening. Typical systems bind a set of
> credentials
> fairly tightly to an AoR (this username password can only be used with
> this AoR
> and its the only username password pair I'll accept for that AoR). In
> that situation,
> returning a 404 would only make sense if you had recently deleted the
> resource
> but hadn't invalidated the credentials yet.
> 
> If the credentials in the second request aren't good, you'll return
> another 401.
> 
> RjS
> 
> On Mar 19, 2007, at 2:01 PM, Martin Weiglhofer wrote:
> 
> Dear all,
> 
> assume, that a user tries to register a contact address on a registrar
> which requires authentication, but the user is not known by the
> registrar. The user sends a REGISTER message (without authentication
> credentials) which is answered with a "401 Unauthorized" response. Now
> the user sends a second REGISTER request which includes an Authorization
> header field. What should be the response of the server to the second
> REGISTER request? Should the registrar again reply with a "401
> Unauthorized", or should the registrar reject the request with a "404
> Not Found"?
> 
> Thanks in advance.
> 
> Regards
> Martin
> 
_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org

-- 
Martin Weiglhofer
Inst. f. Software Technology - Graz University of Technology
Inffeldgasse 16b/II - 8010 Graz - Austria
phone: ++43 316 873 5763
mail: weiglhofer at ist.tugraz.at
web: http://www.ist.tugraz.at/staff/weiglhofer

From danishzaidi54 at yahoo.com  Mon Mar 19 13:24:21 2007
From: danishzaidi54 at yahoo.com (Danish Zaidi)
Date: Mon, 19 Mar 2007 10:24:21 -0700 (PDT)
Subject: [SIPForum-discussion] Need Help with SIP Call Hold
Message-ID: <282483.87232.qm@web90605.mail.mud.yahoo.com>

Dear All

m having trouble with SIP Call Hold in Jain SIP,
didn't find any sample code for Call Hold.

it will be very much helpful for me if any 1 gives me
sample for Call Hold in Jain Sip.


thanx in Advance
Danish Zaidi


 
____________________________________________________________________________________
Don't pick lemons.
See all the new 2007 cars at Yahoo! Autos.
http://autos.yahoo.com/new_cars.html 

From francesco.landolfo at gmail.com  Tue Mar 20 06:23:53 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Tue, 20 Mar 2007 11:23:53 +0100
Subject: [SIPForum-discussion] List of registered user on SIP server
In-Reply-To: <EXCHANGEtpIuPHDXE2s00000589@mail.ingenient.com>
References: <200703171424417035768@wellya.net>
	<EXCHANGEtpIuPHDXE2s00000589@mail.ingenient.com>
Message-ID: <d08f948c0703200323o9352586xf3d42bf05ddad9d9@mail.gmail.com>

I think no.
Why would you try this message? In what scenario would you use it?

2007/3/19, Pavan Jain <pavan.jain at ingenient.com>:
>
>  Is there a message format that I need to send to Server( similar to
> REGISTER)? An example will be helpful.
>
>
>
> Regards,
>
> Pavan
>
>
>  ------------------------------
>
> *From:* wellya [mailto:wellya at wellya.net]
> *Sent:* Saturday, March 17, 2007 2:25 AM
> *To:* Pavan Jain; discussion at sipforum.org
> *Subject:* Re: [SIPForum-discussion] List of registered user on SIP server
>
>
>
> you can query database
>
>
>  ------------------------------
>
> wellya
>
> 2007-03-17
>   ------------------------------
>
> *????* Pavan Jain
>
> *?????* 2007-03-16 22:20:43
>
> *????* discussion at sipforum.org
>
> *???*
>
> *???* [SIPForum-discussion] List of registered user on SIP server
>
>
>
> Hello All:
>
>
>
> How can I get the list of registered users on the SIP server?
>
>
>
> Regards,
>
> Pavan
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>


-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
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From dinis_7 at hotmail.com  Tue Mar 20 06:32:11 2007
From: dinis_7 at hotmail.com (Nelson Dinis)
Date: Tue, 20 Mar 2007 10:32:11 +0000
Subject: [SIPForum-discussion] Types of sessions
Message-ID: <BAY112-F9758F0517CF80D228D832DE750@phx.gbl>

hello all,

I want to establish a session, using SIP, but in my case i don't want that 
the caller (person that send the INVITE) knows what will be the session, 
that need to be decide by the called (person that receive the Invite).

I was tinking sending multiple informacion (multipart body) on the INVITE, 
and then de called decide what will be the session. It possibile to do 
something like this?

It possible use SIP protocol to establish a session that is not media (voip, 
multimedia), for exemple i want to establish a VNC (Virtual Network 
Computing)?

Regards

Nelson Dinis
Portugal

_________________________________________________________________
MSN Hotmail, o maior webmail do Brasil. http://www.hotmail.com


From zeroroot at tmax.co.kr  Tue Mar 20 07:22:18 2007
From: zeroroot at tmax.co.kr (Young-Geun Park)
Date: Tue, 20 Mar 2007 20:22:18 +0900
Subject: [SIPForum-discussion]  how to compute cps(call per second)?
Message-ID: <200703201120.l2KBKelL006065@sipforum.org>

Hi, all

 

I want to know how to compute CPS(call per second) specially with a proxyApp
as follows.

 

UAC          Proxy         UAS

   1(invite)

 ------------>

   2(100 trying)

<-----------

   3(invite)

------------>

4(180 Ringing)

<-----------

5(180 Ringing)

<-----------

6(200 OK)

<-----------

7(200 OK)

<-----------

8(ACK)

------------>

   9(ACK)

------------>

10(BYE)

<-----------

11(BYE)

<-----------

12(200 OK)

------------>

   13(200 OK)

------------>

 

Thanks in advance

 

Regards, 

Park

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From zeroroot at tmax.co.kr  Tue Mar 20 07:31:43 2007
From: zeroroot at tmax.co.kr (Young-Geun Park)
Date: Tue, 20 Mar 2007 20:31:43 +0900
Subject: [SIPForum-discussion] how to compute cps(call per second)?
Message-ID: <200703201130.l2KBU3QY014175@sipforum.org>

and what are there performance factors related to a sip servlet container
that deploys the proxy app?

 

Park

  _____  

From: Young-Geun Park [mailto:zeroroot at tmax.co.kr] 
Sent: Tuesday, March 20, 2007 8:22 PM
To: 'discussion at sipforum.org'
Subject: [SIPForum-discussion] how to compute cps(call per second)?

 

Hi, all

 

I want to know how to compute CPS(call per second) specially with a proxyApp
as follows.

 

UAC          Proxy         UAS

   1(invite)

 ------------>

   2(100 trying)

<-----------

   3(invite)

------------>

4(180 Ringing)

<-----------

5(180 Ringing)

<-----------

6(200 OK)

<-----------

7(200 OK)

<-----------

8(ACK)

------------>

   9(ACK)

------------>

10(BYE)

<-----------

11(BYE)

<-----------

12(200 OK)

------------>

   13(200 OK)

------------>

 

Thanks in advance

 

Regards, 

Park

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From Shanmukharao.Makkapati at airtel.in  Tue Mar 20 09:37:00 2007
From: Shanmukharao.Makkapati at airtel.in (Shanmukharao.Makkapati at airtel.in)
Date: Tue, 20 Mar 2007 19:07:00 +0530
Subject: [SIPForum-discussion] Instant Messaging
Message-ID: <OFFAFFD5B2.88820935-ON652572A4.004A194F-652572A4.004A569E@bharti.com>


Hi,
  I want to get info about a breif sdlc architecture involved in Instant 
messenger using SIP on C, Linux for the developement of whole project.

Can any one help me regard 2 this.



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From pavan.jain at ingenient.com  Tue Mar 20 10:03:49 2007
From: pavan.jain at ingenient.com (Pavan Jain)
Date: Tue, 20 Mar 2007 10:03:49 -0400
Subject: [SIPForum-discussion] List of registered user on SIP server
In-Reply-To: <d08f948c0703200323o9352586xf3d42bf05ddad9d9@mail.gmail.com>
Message-ID: <EXCHANGEy7BXx4XFKpR000005f6@mail.ingenient.com>

Our product needs to know how many users are online (registered) just like we have in IM chat sessions. For that I was wondering my SIP stack can send a Query message from stack to Server, and get a list of registered users. 

 

  _____  

From: Francesco Paolo Landolfo [mailto:francesco.landolfo at gmail.com] 
Sent: Tuesday, March 20, 2007 6:24 AM
To: Pavan Jain
Cc: wellya; discussion at sipforum.org
Subject: Re: [SIPForum-discussion] List of registered user on SIP server

 

I think no. 
Why would you try this message? In what scenario would you use it?

2007/3/19, Pavan Jain <pavan.jain at ingenient.com>: 

Is there a message format that I need to send to Server( similar to REGISTER)? An example will be helpful.

 

Regards,

Pavan

 

  _____  

From: wellya [mailto:wellya at wellya.net] 
Sent: Saturday, March 17, 2007 2:25 AM
To: Pavan Jain; discussion at sipforum.org
Subject: Re: [SIPForum-discussion] List of registered user on SIP server

 

you can query database

  

  _____  

wellya 

2007-03-17 

  _____  

???? Pavan Jain

????? 2007-03-16 22:20:43

???? discussion at sipforum.org 

??? 

??? [SIPForum-discussion] List of registered user on SIP server

  

Hello All:

 

How can I get the list of registered users on the SIP server?

 

Regards,

Pavan


_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org




-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto presto." (C'era una volta in America) 
E adesso so cosa devo fare, devo continuare a respirare perch? domani il sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso) 

-------------- next part --------------
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From francesco.landolfo at gmail.com  Tue Mar 20 10:14:46 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Tue, 20 Mar 2007 15:14:46 +0100
Subject: [SIPForum-discussion] List of registered user on SIP server
In-Reply-To: <EXCHANGEy7BXx4XFKpR000005f6@mail.ingenient.com>
References: <d08f948c0703200323o9352586xf3d42bf05ddad9d9@mail.gmail.com>
	<EXCHANGEy7BXx4XFKpR000005f6@mail.ingenient.com>
Message-ID: <d08f948c0703200714u7cc8db99qfbaf90512494574c@mail.gmail.com>

I don't know how help you. I'm sorry.

Good luck,
Francesco

2007/3/20, Pavan Jain <pavan.jain at ingenient.com>:
>
>  Our product needs to know how many users are online (registered) just
> like we have in IM chat sessions. For that I was wondering my SIP stack can
> send a Query message from stack to Server, and get a list of registered
> users.
>
>
>  ------------------------------
>
> *From:* Francesco Paolo Landolfo [mailto:francesco.landolfo at gmail.com]
> *Sent:* Tuesday, March 20, 2007 6:24 AM
> *To:* Pavan Jain
> *Cc:* wellya; discussion at sipforum.org
> *Subject:* Re: [SIPForum-discussion] List of registered user on SIP server
>
>
>
> I think no.
> Why would you try this message? In what scenario would you use it?
>
> 2007/3/19, Pavan Jain <pavan.jain at ingenient.com>:
>
> Is there a message format that I need to send to Server( similar to
> REGISTER)? An example will be helpful.
>
>
>
> Regards,
>
> Pavan
>
>
>  ------------------------------
>
> *From:* wellya [mailto:wellya at wellya.net]
> *Sent:* Saturday, March 17, 2007 2:25 AM
> *To:* Pavan Jain; discussion at sipforum.org
> *Subject:* Re: [SIPForum-discussion] List of registered user on SIP server
>
>
>
> you can query database
>
>
>  ------------------------------
>
> wellya
>
> 2007-03-17
>   ------------------------------
>
> *????* Pavan Jain
>
> *?????* 2007-03-16 22:20:43
>
> *????* discussion at sipforum.org
>
> *???*
>
> *???* [SIPForum-discussion] List of registered user on SIP server
>
>
>
> Hello All:
>
>
>
> How can I get the list of registered users on the SIP server?
>
>
>
> Regards,
>
> Pavan
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
>
>
> --
> Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
> "Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
> presto." (C'era una volta in America)
> E adesso so cosa devo fare, devo continuare a respirare perch? domani il
> sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
> Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
>



-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
-------------- next part --------------
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From durgani at gmail.com  Tue Mar 20 10:51:58 2007
From: durgani at gmail.com (Prakash Durgani)
Date: Tue, 20 Mar 2007 10:51:58 -0400
Subject: [SIPForum-discussion] Types of sessions
In-Reply-To: <BAY112-F9758F0517CF80D228D832DE750@phx.gbl>
References: <BAY112-F9758F0517CF80D228D832DE750@phx.gbl>
Message-ID: <f6f514070703200751v3530f260s33f689b939d62cf9@mail.gmail.com>

Refer to RFC 3264 on Offer/Answer Model.

On 3/20/07, Nelson Dinis <dinis_7 at hotmail.com> wrote:
>
> hello all,
>
> I want to establish a session, using SIP, but in my case i don't want that
> the caller (person that send the INVITE) knows what will be the session,
> that need to be decide by the called (person that receive the Invite).
>
> I was tinking sending multiple informacion (multipart body) on the INVITE,
> and then de called decide what will be the session. It possibile to do
> something like this?
>
> It possible use SIP protocol to establish a session that is not media
> (voip,
> multimedia), for exemple i want to establish a VNC (Virtual Network
> Computing)?
>
> Regards
>
> Nelson Dinis
> Portugal
>
> _________________________________________________________________
> MSN Hotmail, o maior webmail do Brasil. http://www.hotmail.com
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
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From muraliwind at yahoo.com  Tue Mar 20 13:28:39 2007
From: muraliwind at yahoo.com (Murali ~~)
Date: Tue, 20 Mar 2007 10:28:39 -0700 (PDT)
Subject: [SIPForum-discussion] List of registered user on SIP server
In-Reply-To: <d08f948c0703200714u7cc8db99qfbaf90512494574c@mail.gmail.com>
Message-ID: <874838.94380.qm@web31903.mail.mud.yahoo.com>


Use SUBSCRIBE/NOTIFY!
http://www.ietf.org/rfc/rfc3265.txt

Cheers,
Murali


--- Francesco Paolo Landolfo
<francesco.landolfo at gmail.com> wrote:

> I don't know how help you. I'm sorry.
> 
> Good luck,
> Francesco
> 
> 2007/3/20, Pavan Jain <pavan.jain at ingenient.com>:
> >
> >  Our product needs to know how many users are
> online (registered) just
> > like we have in IM chat sessions. For that I was
> wondering my SIP stack can
> > send a Query message from stack to Server, and get
> a list of registered
> > users.
> >
> >
> >  ------------------------------
> >
> > *From:* Francesco Paolo Landolfo
> [mailto:francesco.landolfo at gmail.com]
> > *Sent:* Tuesday, March 20, 2007 6:24 AM
> > *To:* Pavan Jain
> > *Cc:* wellya; discussion at sipforum.org
> > *Subject:* Re: [SIPForum-discussion] List of
> registered user on SIP server
> >
> >
> >
> > I think no.
> > Why would you try this message? In what scenario
> would you use it?
> >
> > 2007/3/19, Pavan Jain <pavan.jain at ingenient.com>:
> >
> > Is there a message format that I need to send to
> Server( similar to
> > REGISTER)? An example will be helpful.
> >
> >
> >
> > Regards,
> >
> > Pavan
> >
> >
> >  ------------------------------
> >
> > *From:* wellya [mailto:wellya at wellya.net]
> > *Sent:* Saturday, March 17, 2007 2:25 AM
> > *To:* Pavan Jain; discussion at sipforum.org
> > *Subject:* Re: [SIPForum-discussion] List of
> registered user on SIP server
> >
> >
> >
> > you can query database
> >
> >
> >  ------------------------------
> >
> > wellya
> >
> > 2007-03-17
> >   ------------------------------
> >
> > *????????????* Pavan Jain
> >
> > *???????????????* 2007-03-16 22:20:43
> >
> > *????????????* discussion at sipforum.org
> >
> > *?????????*
> >
> > *?????????* [SIPForum-discussion] List of
> registered user on SIP server
> >
> >
> >
> > Hello All:
> >
> >
> >
> > How can I get the list of registered users on the
> SIP server?
> >
> >
> >
> > Regards,
> >
> > Pavan
> >
> >
> > _______________________________________________
> > This is the SIP Forum discussion mailing list
> > TO UNSUBSCRIBE, or edit your delivery options,
> please visit
> > http://sipforum.org/mailman/listinfo/discussion
> > Post to the list at discussion at sipforum.org
> >
> >
> >
> >
> > --
> > Ci?? che facciamo in vita riecheggia
> nell'eternit??...(Il Gladiatore)
> > "Noodles, cos'hai fatto in tutti questi anni?" "
> Sono andato a letto
> > presto." (C'era una volta in America)
> > E adesso so cosa devo fare, devo continuare a
> respirare perch?? domani il
> > sole sorger?? e chiss?? la marea cosa potr??
> portare. (Cast Away)
> > Il progresso! Sempre tardi arriva. (Nuovo Cinema
> Paradiso)
> >
> 
> 
> 
> -- 
> Ci?? che facciamo in vita riecheggia
> nell'eternit??...(Il Gladiatore)
> "Noodles, cos'hai fatto in tutti questi anni?" "
> Sono andato a letto
> presto." (C'era una volta in America)
> E adesso so cosa devo fare, devo continuare a
> respirare perch?? domani il
> sole sorger?? e chiss?? la marea cosa potr??
> portare. (Cast Away)
> Il progresso! Sempre tardi arriva. (Nuovo Cinema
> Paradiso)
> > _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options,
> please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
> 




 
____________________________________________________________________________________
Need Mail bonding?
Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users.
http://answers.yahoo.com/dir/?link=list&sid=396546091

From sakcahalit at hotmail.com  Tue Mar 20 17:13:09 2007
From: sakcahalit at hotmail.com (Halit Sakca)
Date: Tue, 20 Mar 2007 23:13:09 +0200
Subject: [SIPForum-discussion] List of registered user on SIP server
Message-ID: <BAY126-W208EA6B32EB35C7A85835CC750@phx.gbl>

hi Pavan,
In my opinion,
1.) Best way to know how many users are online is querying the database,
2.) On the other side you can use specific debugging tools that can debug the stack(or SIP application server), as far as I know it can turn the number of registered users or call numbers.
 But of course I dont know that your application stack has this tool, program etc...
hopefully, answer was useful :)
 
Halit,


From: pavan.jain at ingenient.comTo: francesco.landolfo at gmail.comDate: Tue, 20 Mar 2007 10:03:49 -0400CC: discussion at sipforum.orgSubject: Re: [SIPForum-discussion] List of registered user on SIP server








Our product needs to know how many users are online (registered) just like we have in IM chat sessions. For that I was wondering my SIP stack can send a Query message from stack to Server, and get a list of registered users. 
 




From: Francesco Paolo Landolfo [mailto:francesco.landolfo at gmail.com] Sent: Tuesday, March 20, 2007 6:24 AMTo: Pavan JainCc: wellya; discussion at sipforum.orgSubject: Re: [SIPForum-discussion] List of registered user on SIP server
 
I think no. Why would you try this message? In what scenario would you use it?

2007/3/19, Pavan Jain <pavan.jain at ingenient.com>: 


Is there a message format that I need to send to Server( similar to REGISTER)? An example will be helpful.
 
Regards,
Pavan
 




From: wellya [mailto:wellya at wellya.net] Sent: Saturday, March 17, 2007 2:25 AMTo: Pavan Jain; discussion at sipforum.orgSubject: Re: [SIPForum-discussion] List of registered user on SIP server
 

you can query database

  




wellya 

2007-03-17 





???? Pavan Jain

????? 2007-03-16 22:20:43

???? discussion at sipforum.org 

??? 

??? [SIPForum-discussion] List of registered user on SIP server

  

Hello All:
 
How can I get the list of registered users on the SIP server?
 
Regards,
Pavan
_______________________________________________This is the SIP Forum discussion mailing listTO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussionPost to the list at discussion at sipforum.org
-- Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto presto." (C'era una volta in America) E adesso so cosa devo fare, devo continuare a respirare perch? domani il sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso) 
_________________________________________________________________
Live.com'u deneyin - h?zl? ve ki?iselle?tirilmi? giri? sayfan?zla istedi?iniz her ?ey tek bir yerde.
http://www.live.com/getstarted
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From rakesh_rcm at yahoo.com  Tue Mar 20 17:31:44 2007
From: rakesh_rcm at yahoo.com (rakesh menon)
Date: Tue, 20 Mar 2007 14:31:44 -0700 (PDT)
Subject: [SIPForum-discussion] Pingtel
Message-ID: <706330.97117.qm@web56612.mail.re3.yahoo.com>

Hi all,

Does anyone work on Pingtel Softphones/Servers?

caio
Rakesh


 
____________________________________________________________________________________
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From jainp1979 at gmail.com  Wed Mar 21 05:34:09 2007
From: jainp1979 at gmail.com (pankaj jain)
Date: Wed, 21 Mar 2007 15:04:09 +0530
Subject: [SIPForum-discussion] doubt in Example-3.7 of RFC 3665
Message-ID: <d41814cf0703210234y3b4c2916o5c6ed48e21328158@mail.gmail.com>

Hi,
I was going through RFC 3665 -- Basic Call Flow Examples: 3.7 - Session with
re-INVITE (IP Address
Change)<http://www.tech-invite.com/SIP/3665/Ti-SIP%203665-3.7%20%28v1.1%29.pdf>
The CSeq header in 1st INVITE (Alice to Bob) is 1
The CSeq header in 2nd INVITE (Bob to Alice) is 14
and The CSeq header in BYE (Alice to Bob) is 2

My questions are:
Shouldn't CSeq be monotonically increasing in a call?
is CSeq similar to TCP seq number where both parties maintain their own
sequence numbers?

-- 
Thanks
Pankaj Jain
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From rjsparks at nostrum.com  Wed Mar 21 06:05:49 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Wed, 21 Mar 2007 11:05:49 +0100
Subject: [SIPForum-discussion] doubt in Example-3.7 of RFC 3665
In-Reply-To: <d41814cf0703210234y3b4c2916o5c6ed48e21328158@mail.gmail.com>
References: <d41814cf0703210234y3b4c2916o5c6ed48e21328158@mail.gmail.com>
Message-ID: <45C80385-434E-4B2C-A647-E75A3C8D01FC@nostrum.com>

The CSeq sequence is scoped to each endpoint (Alice keeps a  
monotonically increasing sequence and Bob keeps a _separate_  
monotonically increasing sequence in this dialog).

RjS

On Mar 21, 2007, at 10:34 AM, pankaj jain wrote:

> Hi,
> I was going through RFC 3665 -- Basic Call Flow Examples: 3.7 -  
> Session with re-INVITE (IP Address Change)
> The CSeq header in 1st INVITE (Alice to Bob) is 1
> The CSeq header in 2nd INVITE (Bob to Alice) is 14
> and The CSeq header in BYE (Alice to Bob) is 2
>
> My questions are:
> Shouldn't CSeq be monotonically increasing in a call?
> is CSeq similar to TCP seq number where both parties maintain their  
> own sequence numbers?
>
> -- 
> Thanks
> Pankaj Jain
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit http:// 
> sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org

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From indresh.singh at siemens.com  Wed Mar 21 09:20:26 2007
From: indresh.singh at siemens.com (Singh, Indresh (SNL US))
Date: Wed, 21 Mar 2007 06:20:26 -0700
Subject: [SIPForum-discussion] doubt in Example-3.7 of RFC 3665
In-Reply-To: <45C80385-434E-4B2C-A647-E75A3C8D01FC@nostrum.com>
Message-ID: <3D80B10873C01D47BEC71C8DE311CF111C7ED2DD@USNWK100MSX.ww017.siemens.net>

Rob is absolutely right. Both parties maintain there own sequence number
as described in the dialog section ( 12.1 ) of RFC-3261
 
A dialog contains certain pieces of state needed for further message

transmissions within the dialog. This state consists of the dialog

ID, a local sequence number (used to order requests from the UA to

its peer), a remote sequence number (used to order requests from its

peer to the UA), a local URI, a remote URI, remote target, a boolean

flag called "secure", and a route set, which is an ordered list of

URIs. The route set is the list of servers that need to be traversed

to send a request to the peer. A dialog can also be in the "early"

state, which occurs when it is created with a provisional response,

and then transition to the "confirmed" state when a 2xx final

response arrives. For other responses, or if no response arrives at

all on that dialog, the early dialog terminates.

Regards,
 
Indresh K Singh

________________________________

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Robert Sparks
Sent: Wednesday, March 21, 2007 6:06 AM
To: pankaj jain
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] doubt in Example-3.7 of RFC 3665


The CSeq sequence is scoped to each endpoint (Alice keeps a
monotonically increasing sequence and Bob keeps a _separate_
monotonically increasing sequence in this dialog). 

RjS

On Mar 21, 2007, at 10:34 AM, pankaj jain wrote:


	Hi,
	I was going through RFC 3665 -- Basic Call Flow Examples: 3.7 -
Session with re-INVITE (IP Address Change)
<http://www.tech-invite.com/SIP/3665/Ti-SIP%203665-3.7%20%28v1.1%29.pdf>

	The CSeq header in 1st INVITE (Alice to Bob) is 1
	The CSeq header in 2nd INVITE (Bob to Alice) is 14
	and The CSeq header in BYE (Alice to Bob) is 2 
	
	My questions are:
	Shouldn't CSeq be monotonically increasing in a call?
	is CSeq similar to TCP seq number where both parties maintain
their own sequence numbers?
	
	-- 
	Thanks
	Pankaj Jain 
	_______________________________________________
	This is the SIP Forum discussion mailing list
	TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
	Post to the list at discussion at sipforum.org


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From Shanmukharao.Makkapati at airtel.in  Wed Mar 21 10:52:05 2007
From: Shanmukharao.Makkapati at airtel.in (Shanmukharao.Makkapati at airtel.in)
Date: Wed, 21 Mar 2007 20:22:05 +0530
Subject: [SIPForum-discussion] IM Using SIP ( Please Help Me )
Message-ID: <OF4F4355CC.256236DF-ON652572A5.005122B9-652572A5.00513511@bharti.com>


Dear Friends,
                  I want to know about Design process in SIP Im using C 
and Linux. If we get Instant messenging project using SIP on C & Linux 
Platform. How can we divide it into modules and aim of each module ( input 
and output )..

Like SIP Parser module. Can you just give me a brief about these 
please.......... Where SDP comes in, Where RTP Comes in, TCP, UDP Use... 
and The design process..., How these modules will be combined to totally 
form a IM Product. please........................please..........


This e-mail and any files transmitted with it are for the sole use of the intended 
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all copies and the original message. Any unauthorized review, use, disclosure, 
dissemination, forwarding, printing or copying of this email or any action taken 
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are unable to exercise control or ensure or guarantee the integrity of/overthe contents 
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From adam.harding2 at ntlworld.com  Wed Mar 21 13:08:42 2007
From: adam.harding2 at ntlworld.com (Adam Harding)
Date: Wed, 21 Mar 2007 17:08:42 +0000
Subject: [SIPForum-discussion] Confused with RTP analysis function in
	Ethereal, please help!
Message-ID: <20070321161151.IQNC17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>

Hi,

This is more of an ethereal/wireshark question but is related to SIP as I am trying to analyse SIP calls:

Please could someone help me as I am quite confused!

What does "Delta" mean in the RTP analysis and how is it calculated?

In the RTP graph analysis, what does the red line indicating "Difference" mean and how is it calculated? 

I thought the "difference" on the graph was giving the Delta results in graph format but the results on the graph are different and lower than the Delta values. 

I am doing my final year project on VOIP and looking into how different network conditions effect the call quality. 

What measurement in wireshark should I be looking at to show the effect that causes bad voice quality due to the variation in packet arrivals at the recieving end? 

Any help would be great as I am very confused!

Thanks.

Adam.


-----------------------------------------
Email sent from www.virginmedia.com/email
Virus-checked using McAfee(R) Software and scanned for spam


From jani_tech_forum at yahoo.com  Wed Mar 21 12:23:05 2007
From: jani_tech_forum at yahoo.com (Janakiraman N)
Date: Wed, 21 Mar 2007 09:23:05 -0700 (PDT)
Subject: [SIPForum-discussion] How to send request to one proxy to another
Message-ID: <300031.93195.qm@web62108.mail.re1.yahoo.com>

Hi ALL,

We are developing conference application using SIP Servlet API (JSR 116 standard).
 
SIP soft phone sending INVITE and we are forwarding that INVITE request to another proxy by using pushRoute() method from SIPServletRequest interface and proxyTo() method from Proxy interface.
 
But SIP soft phone while sending BYE request to close the dialog, remote tag ("To" tag) does not match with 200 OK response which received for INVITE.
 
I can not able to find where the exact problem is located.
 
Please give your valuable suggestion.

Regards,
Janakiraman N


 
____________________________________________________________________________________
Get your own web address.  
Have a HUGE year through Yahoo! Small Business.
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From sukerry at 126.com  Thu Mar 22 00:25:45 2007
From: sukerry at 126.com (sukerry)
Date: Thu, 22 Mar 2007 12:25:45 +0800
Subject: [SIPForum-discussion] how to compute cps(call per second)?
Message-ID: <46020555.01275C.01564@m5-141.126.com>

Young-Geun Park,

Please use sipp and ethreal

======== 2007-03-20 19:31:43 you wrote========

and what are there performance factors related to a sip servlet container that deploys the proxy app?

Park



From: Young-Geun Park [mailto:zeroroot at tmax.co.kr] 
Sent: Tuesday, March 20, 2007 8:22 PM
To: 'discussion at sipforum.org'
Subject: [SIPForum-discussion] how to compute cps(call per second)?

Hi, all

I want to know how to compute CPS(call per second) specially with a proxyApp as follows.

UAC          Proxy         UAS
   1(invite)
 ----------??
   2(100 trying)
<-----------
   3(invite)
----------??
4(180 Ringing)
<-----------
5(180 Ringing)
<-----------
6(200 OK)
<-----------
7(200 OK)
<-----------
8(ACK)
----------??
   9(ACK)
----------??
10(BYE)
<-----------
11(BYE)
<-----------
12(200 OK)
----------??
   13(200 OK)
----------??

Thanks in advance

Regards, 
Park

= = = = = = = = = = = = = = = = = = = = = = 

????????????????????????????sukerry
????????????????????????????sukerry at 126.com
??????????????????????????????2007-03-22
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From pallavim35 at gmail.com  Thu Mar 22 01:10:07 2007
From: pallavim35 at gmail.com (aditi g)
Date: Thu, 22 Mar 2007 10:40:07 +0530
Subject: [SIPForum-discussion] Confused with RTP analysis function in
	Ethereal, please help!
In-Reply-To: <20070321161151.IQNC17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>
References: <20070321161151.IQNC17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>
Message-ID: <63af059d0703212210x78968163l30fcc10e3edc3867@mail.gmail.com>

Hi
Jitter measurement in wireshark shows the effect that causes bad voice
quality due to the variation in packet arrivals at the recieving
end.Acceptable jitter values are upto 40 ms ,if jitter exceeds 40 ms then
poor voice quality can be apprended.

Also delta is time interval between 2 packets which is apprx 20 ms.
If you sort packets in "RTP Analysis: window by Seq no ,you will be able to
see packetization interval
of appx 20 ms which is Delta in this case.

Thanks
Pallavi




On 3/21/07, Adam Harding <adam.harding2 at ntlworld.com> wrote:
>
> Hi,
>
> This is more of an ethereal/wireshark question but is related to SIP as I
> am trying to analyse SIP calls:
>
> Please could someone help me as I am quite confused!
>
> What does "Delta" mean in the RTP analysis and how is it calculated?
>
> In the RTP graph analysis, what does the red line indicating "Difference"
> mean and how is it calculated?
>
> I thought the "difference" on the graph was giving the Delta results in
> graph format but the results on the graph are different and lower than the
> Delta values.
>
> I am doing my final year project on VOIP and looking into how different
> network conditions effect the call quality.
>
> What measurement in wireshark should I be looking at to show the effect
> that causes bad voice quality due to the variation in packet arrivals at the
> recieving end?
>
> Any help would be great as I am very confused!
>
> Thanks.
>
> Adam.
>
>
> -----------------------------------------
> Email sent from www.virginmedia.com/email
> Virus-checked using McAfee(R) Software and scanned for spam
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
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From jainp1979 at gmail.com  Thu Mar 22 01:15:28 2007
From: jainp1979 at gmail.com (pankaj jain)
Date: Thu, 22 Mar 2007 10:45:28 +0530
Subject: [SIPForum-discussion] doubt in Example-3.7 of RFC 3665
In-Reply-To: <3D80B10873C01D47BEC71C8DE311CF111C7ED2DD@USNWK100MSX.ww017.siemens.net>
References: <45C80385-434E-4B2C-A647-E75A3C8D01FC@nostrum.com>
	<3D80B10873C01D47BEC71C8DE311CF111C7ED2DD@USNWK100MSX.ww017.siemens.net>
Message-ID: <d41814cf0703212215h1a3e1582l1f0b0ace5b7efbb6@mail.gmail.com>

Thanks a lot friends
The doubt is cleared now.

On 3/21/07, Singh, Indresh (SNL US) <indresh.singh at siemens.com> wrote:
>
>  Rob is absolutely right. Both parties maintain there own sequence number
> as described in the dialog section ( 12.1 ) of RFC-3261
>
>
> A dialog contains certain pieces of state needed for further message
>
> transmissions within the dialog. This state consists of the dialog
>
> ID, *a local sequence number (used to order requests from the UA to*
>
> *its peer), a remote sequence number (used to order requests from its*
>
> *peer to the UA),* a local URI, a remote URI, remote target, a boolean
>
> flag called "secure", and a route set, which is an ordered list of
>
> URIs. The route set is the list of servers that need to be traversed
>
> to send a request to the peer. A dialog can also be in the "early"
>
> state, which occurs when it is created with a provisional response,
>
> and then transition to the "confirmed" state when a 2xx final
>
> response arrives. For other responses, or if no response arrives at
>
> all on that dialog, the early dialog terminates.
> Regards,
>
> Indresh K Singh
>
>  ------------------------------
> *From:* discussion-bounces at sipforum.org [mailto:
> discussion-bounces at sipforum.org] *On Behalf Of *Robert Sparks
> *Sent:* Wednesday, March 21, 2007 6:06 AM
> *To:* pankaj jain
> *Cc:* discussion at sipforum.org
> *Subject:* Re: [SIPForum-discussion] doubt in Example-3.7 of RFC 3665
>
> The CSeq sequence is scoped to each endpoint (Alice keeps a monotonically
> increasing sequence and Bob keeps a _separate_ monotonically increasing
> sequence in this dialog).
> RjS
>
>  On Mar 21, 2007, at 10:34 AM, pankaj jain wrote:
>
> Hi,
> I was going through RFC 3665 -- Basic Call Flow Examples: 3.7 - Session
> with re-INVITE (IP Address Change)<http://www.tech-invite.com/SIP/3665/Ti-SIP%203665-3.7%20%28v1.1%29.pdf>
> The CSeq header in 1st INVITE (Alice to Bob) is 1
> The CSeq header in 2nd INVITE (Bob to Alice) is 14
> and The CSeq header in BYE (Alice to Bob) is 2
>
> My questions are:
> Shouldn't CSeq be monotonically increasing in a call?
> is CSeq similar to TCP seq number where both parties maintain their own
> sequence numbers?
>
> --
> Thanks
> Pankaj Jain _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
>


-- 
Thanks
Pankaj Jain
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From adam.harding2 at ntlworld.com  Thu Mar 22 08:38:41 2007
From: adam.harding2 at ntlworld.com (Adam Harding)
Date: Thu, 22 Mar 2007 12:38:41 +0000
Subject: [SIPForum-discussion] Simplified algorithm to give some sort of
	E-model result
Message-ID: <20070322123841.XPYB17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>

Hi,

Does anyone know on some sort of basic algorithm that I can put some VOIP analysis results of Jitter, Delay, Packet Loss etc into and get some sort of an estimate of Voice quality, like the R-factor?, but more simple, just to give a rough figure.

I have not got sufficient data or resources to use the proper E-model/MOS/R-Facor, but if there is some sort of free equation I can plug some basic results into just to get some sort of rough figure for voice quality that would be great.

At the moment I am writing a report for my University project and am just giving my opinion on how different factors effect the audio quality.

Just for the benefit of my examiner really, so I have some sort of numerical figure to use to compare the my results, rather than just my opinion of the audio quality.


Thanks,

Adam Harding

-----------------------------------------
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From gargashish11 at gmail.com  Thu Mar 22 10:03:34 2007
From: gargashish11 at gmail.com (Ashish Garg)
Date: Thu, 22 Mar 2007 19:33:34 +0530
Subject: [SIPForum-discussion] Query on SIPp
Message-ID: <f11c9280703220703o3cb85e8cv61001cb1c31d2e39@mail.gmail.com>

Hi,
 Does anyone know while using RTP echo feature of *SIPp* why the tool
echo the RTP/UDP packets back to the sender coming on the port specfied +2?

*Statements as in Documentation of SIPp:*

*The "RTP echo" feature allows SIPp to listen to one or two local IP address
and port (specified using -mi and -mp command line parameters) for RTP
media. Everything that is received on this address/port is echoed back to
the sender. *

*RTP/UDP packets coming on this port + 2 are also echoed to their sender
(used for sound and video echo).*

** Thanks

Ashish Garg
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From varlei.knupp at siemens.com  Thu Mar 22 11:30:32 2007
From: varlei.knupp at siemens.com (Knupp, Varlei Fernandes)
Date: Thu, 22 Mar 2007 12:30:32 -0300
Subject: [SIPForum-discussion] doubt about "contact field"
Message-ID: <52D77941ACCDE243874A51D2E17489540207F1E7@SAO1015V.ww101.siemens.net>



Hi to all,

I have a doubt about "contact field".

The proxy send a "INVITE" message =INVITE sip:33133010 at 192.168.2.100:5060 SIP/2.0 to Gateway

The gateway send  a "session progress" message to proxy, but this message has contact field = Contact: <sip:1100 at 192.168.2.100>

The User disconnect the call.

The proxy send a "CANCEL" message to gateway = CANCEL sip:1100 at 192.168.2.100:5060 SIP/2.0 

So the gateway reject the message.

Who is correct?
- The Gateway because reject the call because the proxy changed the header in the message CANCEL
- The Proxy because when the gateway sent contact field = <sip:1100 at 192.168.2.100> the header CANCEL message now is  CANCEL sip:1100 at 192.168.2.100:5060 SIP/2.0.

thanks a lot


Varlei Fernandes Knupp

SIEMENS Communications

Customer Services Enterprise

RCC S?o Paulo - Technical Support

Tel.: +55 11 3817 2659

Fax: +55 11 3817 2613

varlei.knupp at siemens.com <blocked::mailto:varlei.knupp at siemens.com>




________________________________________
The information contained in this e-mail is for the exclusive use of the intended recipient(s) and may be confidential, proprietary, and/or legally privileged.  Inadvertent disclosure of this message does not constitute a waiver of any privilege.  If you receive this message in error, please do not directly or indirectly use, print, copy, forward, or disclose any part of this message.  Please also delete this e-mail and all copies and notify the sender. Thank you. For alternate languages please go to http://www.siemens.com.ar/disclaimer/
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From Shanmukharao.Makkapati at airtel.in  Thu Mar 22 13:37:44 2007
From: Shanmukharao.Makkapati at airtel.in (Shanmukharao.Makkapati at airtel.in)
Date: Thu, 22 Mar 2007 23:07:44 +0530
Subject: [SIPForum-discussion] IM Design
Message-ID: <OF0E64B733.DF99E92A-ON652572A6.00608D89-652572A6.00605E49@bharti.com>


Hi to all,

When we design IM using SIP 

The modules may be, 

1.       we may have to have parser module (to parse incmonibg/outgoing 
msges),
2.       core call engine (in case if you want VOIP telephony),
3.       authentication modules to check the validity of the accounts,
4.       network module for data transfer (UDP for IM) and
5.       presence module to provide status of the user (online/offline)..

How these modules will be structered in a sequence to design IM.

Can any one provide me the aritecture of the same in breif please....





This e-mail and any files transmitted with it are for the sole use of the intended 
recipient(s) and may contain confidential and privileged information. If you are 
not the intended recipient, please contact the sender by reply e-mail and destroy 
all copies and the original message. Any unauthorized review, use, disclosure, 
dissemination, forwarding, printing or copying of this email or any action taken 
in reliance on this e-mail is strictly prohibited and may be unlawful.The recipient 
acknowledges that Bharti Airtel Limited or its subsidiaries and associated companies
(collectively "Bharti Airtel Limited"), 
are unable to exercise control or ensure or guarantee the integrity of/overthe contents 
of the information contained in e-mail transmissions and further acknowledges that any views 
expressed in this message are those of the individual sender and no binding nature of the message shall be implied or assumed 
unless the sender does so expressly with due authority of BHARTI AIRTEL LTD.. Before opening any attachments 
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From indresh.singh at siemens.com  Thu Mar 22 16:16:24 2007
From: indresh.singh at siemens.com (Singh, Indresh (SNL US))
Date: Thu, 22 Mar 2007 13:16:24 -0700
Subject: [SIPForum-discussion] doubt about "contact field"
In-Reply-To: <52D77941ACCDE243874A51D2E17489540207F1E7@SAO1015V.ww101.siemens.net>
Message-ID: <3D80B10873C01D47BEC71C8DE311CF111CB117AB@USNWK100MSX.ww017.siemens.net>

CANCEL message has to be exactly same as INVITE request and also should be sent to the same host port as original INVITE. So in this case gateway correctly rejects the message.

________________________________

From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Knupp, Varlei Fernandes
Sent: Thursday, March 22, 2007 11:31 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] doubt about "contact field"


 
Hi to all,
 
I have a doubt about "contact field".
 
The proxy send a "INVITE" message =INVITE sip:33133010 at 192.168.2.100:5060 SIP/2.0 to Gateway
 
The gateway send  a "session progress" message to proxy, but this message has contact field = Contact: <sip:1100 at 192.168.2.100>
 
The User disconnect the call.
 
The proxy send a "CANCEL" message to gateway = CANCEL sip:1100 at 192.168.2.100:5060 SIP/2.0  
 
So the gateway reject the message.
 
Who is correct? 
- The Gateway because reject the call because the proxy changed the header in the message CANCEL
- The Proxy because when the gateway sent contact field = <sip:1100 at 192.168.2.100> the header CANCEL message now is  CANCEL sip:1100 at 192.168.2.100:5060 SIP/2.0.
 
thanks a lot
 
 
Varlei Fernandes Knupp 

SIEMENS Communications

Customer Services Enterprise

RCC S?o Paulo - Technical Support

Tel.: +55 11 3817 2659 

Fax: +55 11 3817 2613

varlei.knupp at siemens.com <blocked::mailto:varlei.knupp at siemens.com> 

 
________________________________________
The information contained in this e-mail is for the exclusive use of the intended recipient(s) and may be confidential, proprietary, and/or legally privileged. Inadvertent disclosure of this message does not constitute a waiver of any privilege. If you receive this message in error, please do not directly or indirectly use, print, copy, forward, or disclose any part of this message. Please also delete this e-mail and all copies and notify the sender. Thank you. For alternate languages please go to http://www.siemens.com.ar/disclaimer/
	
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From varlei.knupp at siemens.com  Thu Mar 22 16:19:29 2007
From: varlei.knupp at siemens.com (Knupp, Varlei Fernandes)
Date: Thu, 22 Mar 2007 17:19:29 -0300
Subject: [SIPForum-discussion] doubt about "contact field"
In-Reply-To: <3D80B10873C01D47BEC71C8DE311CF111CB117AB@USNWK100MSX.ww017.siemens.net>
References: <52D77941ACCDE243874A51D2E17489540207F1E7@SAO1015V.ww101.siemens.net>
	<3D80B10873C01D47BEC71C8DE311CF111CB117AB@USNWK100MSX.ww017.siemens.net>
Message-ID: <52D77941ACCDE243874A51D2E17489540207F54B@SAO1015V.ww101.siemens.net>


Thanks a lot for your help.:-)


Varlei Fernandes Knupp

SIEMENS Communications

Customer Services Enterprise

RCC S?o Paulo - Technical Support

Tel.: +55 11 3817 2659

Fax: +55 11 3817 2613

varlei.knupp at siemens.com <blocked::mailto:varlei.knupp at siemens.com>



________________________________

From: Singh, Indresh (SNL US)
Sent: quinta-feira, 22 de mar?o de 2007 17:16
To: Knupp, Varlei Fernandes; discussion at sipforum.org
Subject: RE: [SIPForum-discussion] doubt about "contact field"


CANCEL message has to be exactly same as INVITE request and also should be sent to the same host port as original INVITE. So in this case gateway correctly rejects the message.

________________________________

From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Knupp, Varlei Fernandes
Sent: Thursday, March 22, 2007 11:31 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] doubt about "contact field"



Hi to all,

I have a doubt about "contact field".

The proxy send a "INVITE" message =INVITE sip:33133010 at 192.168.2.100:5060 SIP/2.0 to Gateway

The gateway send  a "session progress" message to proxy, but this message has contact field = Contact: <sip:1100 at 192.168.2.100>

The User disconnect the call.

The proxy send a "CANCEL" message to gateway = CANCEL sip:1100 at 192.168.2.100:5060 SIP/2.0 

So the gateway reject the message.

Who is correct?
- The Gateway because reject the call because the proxy changed the header in the message CANCEL
- The Proxy because when the gateway sent contact field = <sip:1100 at 192.168.2.100> the header CANCEL message now is  CANCEL sip:1100 at 192.168.2.100:5060 SIP/2.0.

thanks a lot


Varlei Fernandes Knupp

SIEMENS Communications

Customer Services Enterprise

RCC S?o Paulo - Technical Support

Tel.: +55 11 3817 2659

Fax: +55 11 3817 2613

varlei.knupp at siemens.com <blocked::mailto:varlei.knupp at siemens.com>


________________________________________
The information contained in this e-mail is for the exclusive use of the intended recipient(s) and may be confidential, proprietary, and/or legally privileged. Inadvertent disclosure of this message does not constitute a waiver of any privilege. If you receive this message in error, please do not directly or indirectly use, print, copy, forward, or disclose any part of this message. Please also delete this e-mail and all copies and notify the sender. Thank you. For alternate languages please go to http://www.siemens.com.ar/disclaimer/



________________________________________
The information contained in this e-mail is for the exclusive use of the intended recipient(s) and may be confidential, proprietary, and/or legally privileged.  Inadvertent disclosure of this message does not constitute a waiver of any privilege.  If you receive this message in error, please do not directly or indirectly use, print, copy, forward, or disclose any part of this message.  Please also delete this e-mail and all copies and notify the sender. Thank you. For alternate languages please go to http://www.siemens.com.ar/disclaimer/
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From cmarzotta at gmail.com  Thu Mar 22 16:55:38 2007
From: cmarzotta at gmail.com (Claudio Marzotta)
Date: Thu, 22 Mar 2007 17:55:38 -0300
Subject: [SIPForum-discussion] contact field
In-Reply-To: <3D80B10873C01D47BEC71C8DE311CF111CB117AB@USNWK100MSX.ww017.siemens.net>
References: <52D77941ACCDE243874A51D2E17489540207F1E7@SAO1015V.ww101.siemens.net>
	<3D80B10873C01D47BEC71C8DE311CF111CB117AB@USNWK100MSX.ww017.siemens.net>
Message-ID: <005501c76cc4$73ce05a0$080000c8@bcdros.com>

 

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From danishzaidi54 at yahoo.com  Thu Mar 22 19:50:10 2007
From: danishzaidi54 at yahoo.com (Danish Zaidi)
Date: Thu, 22 Mar 2007 16:50:10 -0700 (PDT)
Subject: [SIPForum-discussion] am i missing something with call hold
Message-ID: <290796.4343.qm@web90604.mail.mud.yahoo.com>

Hello 

the Hold Event is sent like the INVITE the difference is only attribute is sendonly

if m correct then why its not working, this sendHold Code works if i use it for Invite Purpose only...

public void sendHold()
    {        
        try
        {
            toUser = "1112";
            outgoingCall = true;
            rtpConnection = new RtpConnection();
            localRtpPort = rtpConnection.inizialize(localRtpStartPort, localRtpEndPort, bufferLenght, minimumThreshold, enabledThreshold, packetSize);
            System.out.print("Sending INVITE...  ");
            reqUnauthInvite = null;
            reqAuthInvite = null;
            SipURI requestURI = addressFactory.createSipURI(toUser, serverIpPort);
            SipURI toAddress = addressFactory.createSipURI(toUser, serverIp);
            System.out.println("Server IP In Client.java is "+serverIp);
            Address toNameAddress = addressFactory.createAddress(toAddress);
            ToHeader toHeader = headerFactory.createToHeader(toNameAddress, null);
            System.out.println("IPAddress In Client.java is "+sipStack.getIPAddress());
            SipURI fromAddress = addressFactory.createSipURI(username, sipStack.getIPAddress());
            Address fromNameAddress = addressFactory.createAddress(fromAddress);
            FromHeader fromHeader = headerFactory.createFromHeader(fromNameAddress, userTag);
            ArrayList viaHeaders = new ArrayList();
            javax.sip.header.ViaHeader viaHeader = headerFactory.createViaHeader(sipStack.getIPAddress(), sipProvider.getListeningPoint().getPort(), transportProt, null);
            viaHeaders.add(viaHeader);
            CallIdHeader callIdHeader = cldTemp;
            CSeqHeader cSeqHeader = headerFactory.createCSeqHeader(1, "INVITE");
            MaxForwardsHeader maxForwards = headerFactory.createMaxForwardsHeader(70);
            Request request = messageFactory.createRequest(requestURI, "INVITE", callIdHeader, cSeqHeader, fromHeader, toHeader, viaHeaders, maxForwards);
            request.addHeader(contactHeader);
            ContentTypeHeader contentTypeHeader = headerFactory.createContentTypeHeader("application", "sdp");
            String myAddress = Globals.addr.getHostAddress();            
            String string1 = " RTP/AVP";
            String string2 = "";
            int i=0;
            //for(int i = 0; i < codecListModel.getSize(); i++)
            {
                if(String.valueOf(codecListModel.elementAt(i)).equals(" PCMU/8000 "))
                {
                    string1 = (new StringBuilder()).append(string1).append(" 0").toString();
                    string2 = (new StringBuilder()).append(string2).append("a=rtpmap:0 PCMU/8000\r\n").toString();
                    //continue;
                }
                if(String.valueOf(codecListModel.elementAt(i)).equals(" GSM/8000 "))
                {
                    string1 = (new StringBuilder()).append(string1).append(" 3").toString();
                    string2 = (new StringBuilder()).append(string2).append("a=rtpmap:3 GSM/8000\r\n").toString();
                    //continue;
                }
                if(String.valueOf(codecListModel.elementAt(i)).equals(" G723/8000 "))
                {
                    string1 = (new StringBuilder()).append(string1).append(" 4").toString();
                    string2 = (new StringBuilder()).append(string2).append("a=rtpmap:4 G723/8000\r\n").toString();
                    //continue;
                }
                if(String.valueOf(codecListModel.elementAt(i)).equals(" DVI4/8000 "))
                {
                    string1 = (new StringBuilder()).append(string1).append(" 5").toString();
                    string2 = (new StringBuilder()).append(string2).append("a=rtpmap:5 DVI4/8000\r\n").toString();
                }
            }

            string2+="a=sendonly";
            string1 = (new StringBuilder()).append(string1).append("\r\n").toString();
            String sdpData = (new StringBuilder()).append("v=0\r\no=4855 13760799956958020 13760799956958020 IN IP4  ").append(myAddress).append("\r\n").append("s=Session SDP\r\n").append("c=IN IP4  ").append(myAddress).append("\r\n").append("t=0 0\r\n").append("m=audio ").append(localRtpPort).append(string1).append(string2).toString();
            byte contents[] = sdpData.getBytes();
            request.setContent(contents, contentTypeHeader);
            request.addHeader(userAgentHeader);
            javax.sip.header.Header callInfoHeader = headerFactory.createHeader("Call-Info", "<http://www.kleis.it>");
            request.addHeader(callInfoHeader);
            inviteTid = sipProvider.getNewClientTransaction(request);
            inviteTid.sendRequest();
            dialog = inviteTid.getDialog();
            reqUnauthInvite = request;
            System.out.println("DONE");
        }
        catch(Exception ex)
        {
            System.out.println(ex.getMessage());
            ex.printStackTrace();
        }
        return;
    }


sorrry about the indentation mistakes

but plzz help me with the SIP Call HOLD

thanx in advance

 
---------------------------------
Bored stiff? Loosen up...
Download and play hundreds of games for free on Yahoo! Games.
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From peter324.kim at samsung.com  Thu Mar 22 20:27:45 2007
From: peter324.kim at samsung.com (HYUNGON KIM)
Date: Fri, 23 Mar 2007 00:27:45 +0000 (GMT)
Subject: [SIPForum-discussion] Unscribe
Message-ID: <0JFB00IP2YM9VE@ms5.samsung.com>

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From peter324.kim at samsung.com  Thu Mar 22 20:29:46 2007
From: peter324.kim at samsung.com (HYUNGON KIM)
Date: Fri, 23 Mar 2007 00:29:46 +0000 (GMT)
Subject: [SIPForum-discussion] Unsubscribe
Message-ID: <0JFB00MH8YPMQW@ms5.samsung.com>

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From ramank24 at gmail.com  Fri Mar 23 00:13:05 2007
From: ramank24 at gmail.com (raman kumar)
Date: Fri, 23 Mar 2007 04:13:05 +0000
Subject: [SIPForum-discussion] how to compute cps(call per second)?
In-Reply-To: <46020555.01275C.01564@m5-141.126.com>
References: <46020555.01275C.01564@m5-141.126.com>
Message-ID: <67002eb30703222113h242732d6n613e36cea0331719@mail.gmail.com>

caps defines the load capacity of sip proxy.
It is number of INVITE message it can handle without with reply of
TRYING 100 message without droping any call at same time( call set up
time may be different )

eg if your SIP proxy is capable of responding to 100 INVITE message
per second ( think a phone is sending 100 INVITE per second and is
capable of making 100 connections )

These things can also be tested using sipp tool

On 22/03/07, sukerry <sukerry at 126.com> wrote:
> Young-Geun Park,
>
> Please use sipp and ethreal
>
> ======== 2007-03-20 19:31:43 you wrote========
>
> and what are there performance factors related to a sip servlet container
> that deploys the proxy app?
>
> Park
>
>
>
> From: Young-Geun Park [mailto:zeroroot at tmax.co.kr]
> Sent: Tuesday, March 20, 2007 8:22 PM
> To: 'discussion at sipforum.org'
> Subject: [SIPForum-discussion] how to compute cps(call per second)?
>
> Hi, all
>
> I want to know how to compute CPS(call per second) specially with a proxyApp
> as follows.
>
> UAC          Proxy         UAS
>    1(invite)
>  ----------??
>    2(100 trying)
> <-----------
>    3(invite)
> ----------??
> 4(180 Ringing)
> <-----------
> 5(180 Ringing)
> <-----------
> 6(200 OK)
> <-----------
> 7(200 OK)
> <-----------
> 8(ACK)
> ----------??
>    9(ACK)
> ----------??
> 10(BYE)
> <-----------
> 11(BYE)
> <-----------
> 12(200 OK)
> ----------??
>    13(200 OK)
> ----------??
>
> Thanks in advance
>
> Regards,
> Park
>
> = = = = = = = = = = = = = = = = = = = = = =
>
> ????????????????????????????sukerry
> ????????????????????????????sukerry at 126.com
> ??????????????????????????????2007-03-22
>


From skp10_9559 at yahoo.co.in  Fri Mar 23 00:32:13 2007
From: skp10_9559 at yahoo.co.in (santosh patra)
Date: Fri, 23 Mar 2007 10:02:13 +0530 (IST)
Subject: [SIPForum-discussion] Add in Contact List
Message-ID: <653482.24520.qm@web8812.mail.in.yahoo.com>




		
__________________________________________________________
Yahoo! India Answers: Share what you know. Learn something new
http://in.answers.yahoo.com/
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From vtsoukanov at jnetx.ru  Fri Mar 23 03:53:25 2007
From: vtsoukanov at jnetx.ru (Victor Tsoukanov)
Date: Fri, 23 Mar 2007 10:53:25 +0300
Subject: [SIPForum-discussion] IM Design
References: <OF0E64B733.DF99E92A-ON652572A6.00608D89-652572A6.00605E49@bharti.com>
Message-ID: <00ba01c76d20$57276890$d800a8c0@jnetx.ru>


----- Original Message ----- 
From: Shanmukharao.Makkapati at airtel.in
To: discussion at sipforum.org
Sent: Thursday, March 22, 2007 8:37 PM
Subject: [SIPForum-discussion] IM Design



Hi to all,

When we design IM using SIP

The modules may be,

1.         we may have to have parser module (to parse incmonibg/outgoing
msges),
2.         core call engine (in case if you want VOIP telephony),
3.         authentication modules to check the validity of the accounts,
4.         network module for data transfer (UDP for IM) and
5.         presence module to provide status of the user (online/offline)..

How these modules will be structered in a sequence to design IM.

Can any one provide me the aritecture of the same in breif please....

Hi

Look at SIP servlet API. First and fourth items already implemetnted in SIP
servlet container, other ones can be implemented with different servlets.
For example, second item (as I understand it is call control) - can be
implemeted with B2B servlet, authentication module and presence module - is
a simple proxy servlets. All this stuff can be collected in one application
with simple dispatcher servlet or controller servlet (there are a lot of
suitable examples for HTTP servlets).


From rjsparks at nostrum.com  Fri Mar 23 04:08:06 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Fri, 23 Mar 2007 09:08:06 +0100
Subject: [SIPForum-discussion] SIPit 20 registration closes March 30
Message-ID: <1CC1DF4D-00A0-4DE8-8E63-610B80C488BC@nostrum.com>

SIPit 20 registration closes one week from today (March 30).

The event will be in Antwerp, Belgium hosted by Alcatel-Lucent.

If you are planning to attend, but have not registered, please do so  
now.
Information and a link to the registration site is available at
http://www.sipit.net

RjS

From pallavim35 at gmail.com  Fri Mar 23 04:22:10 2007
From: pallavim35 at gmail.com (aditi g)
Date: Fri, 23 Mar 2007 13:52:10 +0530
Subject: [SIPForum-discussion] Unsubscribe
In-Reply-To: <0JFB00MH8YPMQW@ms5.samsung.com>
References: <0JFB00MH8YPMQW@ms5.samsung.com>
Message-ID: <63af059d0703230122w5f696d01u528b3d3d3d82a840@mail.gmail.com>

Hello

Here when you are sending Invite to hold the call,it is Reinvite that you
have to send.
So from code ,i could see that you are creating new client
transaction.Thisis wrong as for hold ,
you have send REinvite with same sip header as orginal invite but with
different SDP where you can
specify a= sendonly.You do not have to create new transaction.
Thanks


On 3/23/07, HYUNGON KIM <peter324.kim at samsung.com> wrote:
>
> Unsubscribe
>
>
>
>
>
>
> ------- *Original Message* -------
> *Sender* : Danish Zaidi<danishzaidi54 at yahoo.com>
> *Date* : 2007-03-23 08:50
> *Title* : [SIPForum-discussion] am i missing something with call hold
>
> Hello
>
> the Hold Event is sent like the INVITE the difference is only attribute is
> sendonly
>
> if m correct then why its not working, this sendHold Code works if i use
> it for Invite Purpose only...
>
> public void sendHold()
>     {
>         try
>         {
>             toUser = "1112";
>             outgoingCall = true;
>             rtpConnection = new RtpConnection();
>             localRtpPort = rtpConnection.inizialize(localRtpStartPort,
> localRtpEndPort, bufferLenght, minimumThreshold, enabledThreshold,
> packetSize);
>             System.out.print("Sending INVITE...  ");
>             reqUnauthInvite = null;
>             reqAuthInvite = null;
>             SipURI requestURI = addressFactory.createSipURI(toUser,
> serverIpPort);
>             SipURI toAddress = addressFactory.createSipURI(toUser,
> serverIp);
>             System.out.println("Server IP In Client.java is "+serverIp);
>             Address toNameAddress = addressFactory.createAddress
> (toAddress);
>             ToHeader toHeader = headerFactory.createToHeader(toNameAddress,
> null);
>             System.out.println("IPAddress In Client.java is
> "+sipStack.getIPAddress());
>             SipURI fromAddress = addressFactory.createSipURI(username,
> sipStack.getIPAddress());
>             Address fromNameAddress = addressFactory.createAddress
> (fromAddress);
>             FromHeader fromHeader = headerFactory.createFromHeader(fromNameAddress,
> userTag);
>             ArrayList viaHeaders = new ArrayList();
>             javax.sip.header.ViaHeader viaHeader =
> headerFactory.createViaHeader(sipStack.getIPAddress(),
> sipProvider.getListeningPoint().getPort(), transportProt, null);
>             viaHeaders.add(viaHeader);
>             CallIdHeader callIdHeader = cldTemp;
>             CSeqHeader cSeqHeader = headerFactory.createCSeqHeader(1,
> "INVITE");
>             MaxForwardsHeader maxForwards =
> headerFactory.createMaxForwardsHeader(70);
>             Request request = messageFactory.createRequest(requestURI,
> "INVITE", callIdHeader, cSeqHeader, fromHeader, toHeader, viaHeaders,
> maxForwards);
>             request.addHeader(contactHeader);
>             ContentTypeHeader contentTypeHeader =
> headerFactory.createContentTypeHeader("application", "sdp");
>             String myAddress = Globals.addr.getHostAddress();
>             String string1 = " RTP/AVP";
>             String string2 = "";
>             int i=0;
>             //for(int i = 0; i < codecListModel.getSize(); i++)
>             {
>                 if(String.valueOf(codecListModel.elementAt(i)).equals("
> PCMU/8000 "))
>                 {
>                     string1 = (new
> StringBuilder()).append(string1).append(" 0").toString();
>                     string2 = (new
> StringBuilder()).append(string2).append("a=rtpmap:0
> PCMU/8000\r\n").toString();
>                     //continue;
>                 }
>                 if(String.valueOf(codecListModel.elementAt(i)).equals("
> GSM/8000 "))
>                 {
>                     string1 = (new
> StringBuilder()).append(string1).append(" 3").toString();
>                     string2 = (new
> StringBuilder()).append(string2).append("a=rtpmap:3
> GSM/8000\r\n").toString();
>                     //continue;
>                 }
>                 if(String.valueOf(codecListModel.elementAt(i)).equals("
> G723/8000 "))
>                 {
>                     string1 = (new
> StringBuilder()).append(string1).append(" 4").toString();
>                     string2 = (new
> StringBuilder()).append(string2).append("a=rtpmap:4
> G723/8000\r\n").toString();
>                     //continue;
>                 }
>                 if(String.valueOf(codecListModel.elementAt(i)).equals("
> DVI4/8000 "))
>                 {
>                     string1 = (new
> StringBuilder()).append(string1).append(" 5").toString();
>                     string2 = (new
> StringBuilder()).append(string2).append("a=rtpmap:5
> DVI4/8000\r\n").toString();
>                 }
>             }
>
>             string2+="a=sendonly";
>             string1 = (new
> StringBuilder()).append(string1).append("\r\n").toString();
>             String sdpData = (new StringBuilder()).append("v=0\r\no=4855
> 13760799956958020 13760799956958020 IN IP4
> ").append(myAddress).append("\r\n").append("s=Session SDP\r\n").append("c=IN
> IP4  ").append(myAddress).append("\r\n").append("t=0 0\r\n").append("m=audio
> ").append(localRtpPort).append(string1).append(string2).toString();
>             byte contents[] = sdpData.getBytes();
>             request.setContent(contents, contentTypeHeader);
>             request.addHeader(userAgentHeader);
>             javax.sip.header.Header callInfoHeader =
> headerFactory.createHeader("Call-Info", "<http://www.kleis.it>");
>             request.addHeader(callInfoHeader);
>             inviteTid = sipProvider.getNewClientTransaction(request);
>             inviteTid.sendRequest();
>             dialog = inviteTid.getDialog();
>             reqUnauthInvite = request;
>             System.out.println("DONE");
>         }
>         catch(Exception ex)
>         {
>             System.out.println(ex.getMessage());
>             ex.printStackTrace();
>         }
>         return;
>     }
>
>
> sorrry about the indentation mistakes
>
> but plzz help me with the SIP Call HOLD
>
> thanx in advance
>
> ------------------------------
> Bored stiff? <http://us.rd.yahoo.com/evt=49935/*http://games.yahoo.com>Loosen up...
> Download and play hundreds of games for free<http://us.rd.yahoo.com/evt=49935/*http://games.yahoo.com>on Yahoo! Games.
>
>
>
>
>        <http://www.samsungnetworks.co.kr/>   ??? (KIM HYUN GON) ??
> ?????? ??3? Tel. 070-7015-0442 Fax.070-7015-5678 M.P  017-642-1518
> peter324.kim at samsung.com    <http://www.wyz070.co.kr/>
>
>
>
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
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From yong2.chen at siemens.com  Fri Mar 23 04:22:24 2007
From: yong2.chen at siemens.com (Chen, Yong SNLB PEK)
Date: Fri, 23 Mar 2007 16:22:24 +0800
Subject: [SIPForum-discussion] Unscribe
Message-ID: <FD353C9F0D65454D911AFD6D3669ADC405693633@PEKW934A.cn001.siemens.net>

Unscribe
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From Alois.Komenda at R-KOM.de  Fri Mar 23 05:42:53 2007
From: Alois.Komenda at R-KOM.de (Komenda Alois)
Date: Fri, 23 Mar 2007 10:42:53 +0100
Subject: [SIPForum-discussion] Experience with SIP Application Servers
Message-ID: <2A5386A266D8DB11A4AE0090275130BC06A17D@ffserver>

Hello,

has anyone of you experience on working with one of these Application
Servers: ApexVoice OmniVox3D, Aricent SIP AS, BEA WebLogic SIP, Pactolus
RapidFLEX, Ubiquity SIP AS?

Can you tell me about strength and weaknesses of these Servers, supported
features and developing new services?

Best regards

Alois Komenda


From Rakesh.Hooda at aricent.com  Fri Mar 23 06:09:27 2007
From: Rakesh.Hooda at aricent.com (Rakesh Hooda)
Date: Fri, 23 Mar 2007 15:39:27 +0530
Subject: [SIPForum-discussion] Experience with SIP Application Servers
Message-ID: <OFEB66B940.6DC47CF1-ON652572A7.0037CC31-652572A7.0037CC35@flextronicssoftware.com>

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From francesco.landolfo at gmail.com  Fri Mar 23 06:59:07 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Fri, 23 Mar 2007 11:59:07 +0100
Subject: [SIPForum-discussion] How to implement a chat M2M
Message-ID: <d08f948c0703230359h55727018x62fe914d05c96de1@mail.gmail.com>

Hi,
I have implemented a chat P2P using SIP and the MESSAGE message.
Pratically, if Pippo want to send an instant message to Pluto, Pippo sends a
MESSAGE to a Server, this Server convert the Pluto Sip Uri in Pluto Care of
Address and send it to Pluto. This is in agreement with rfc3261.

Now I'd like to implement a chat M2M using SIP, but I have some doubt about
flow and messages that I have to use.

I think that there is one possible scene:

   - A client send a MESSAGE to a Server that dispatchs this message to
   all chat partecipant.

Now, how can I implement this feature?
Have someone some other idea?

Thanks,
Francesco

-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
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From qt.kiran at gmail.com  Fri Mar 23 07:19:10 2007
From: qt.kiran at gmail.com (kiran chakkilam)
Date: Fri, 23 Mar 2007 16:49:10 +0530
Subject: [SIPForum-discussion] pls help me in this basic scenario
Message-ID: <e0646ce70703230419j7910efb4r389f27d192cea550@mail.gmail.com>

Hi everybody,


I am currently working on the sipp1.1 . i want to test proxy server(System
Under test). so i am simulating UAC and UAS using sipp. i will

send invite messge from UAC ,it will hit proxy(SUT).Proxy responds with 100
trying response back to UAC with all headers in short format

then my UAC is terminated.It's showing an error  No valid call-id. so I have
some doubts in sipp.

1. whether sipp accepts headers in shorcode format.

2. why it's terminating after receiving the 100 trying response.


so kindly help me in this scenario,it's very urgent for me


Thanks & Regards

ch.kiran
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From pallavim35 at gmail.com  Fri Mar 23 04:33:22 2007
From: pallavim35 at gmail.com (aditi g)
Date: Fri, 23 Mar 2007 14:03:22 +0530
Subject: [SIPForum-discussion] Simplified algorithm to give some sort of
	E-model result
In-Reply-To: <20070322123841.XPYB17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>
References: <20070322123841.XPYB17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>
Message-ID: <63af059d0703230133t3dd262aaibe8f50302a16bc14@mail.gmail.com>

Hello,
I am sending document as attachment that shows calcualtion of traffic
metrics.
Thanks
Pallavi


On 3/22/07, Adam Harding <adam.harding2 at ntlworld.com> wrote:
>
> Hi,
>
> Does anyone know on some sort of basic algorithm that I can put some VOIP
> analysis results of Jitter, Delay, Packet Loss etc into and get some sort of
> an estimate of Voice quality, like the R-factor?, but more simple, just to
> give a rough figure.
>
> I have not got sufficient data or resources to use the proper
> E-model/MOS/R-Facor, but if there is some sort of free equation I can plug
> some basic results into just to get some sort of rough figure for voice
> quality that would be great.
>
> At the moment I am writing a report for my University project and am just
> giving my opinion on how different factors effect the audio quality.
>
> Just for the benefit of my examiner really, so I have some sort of
> numerical figure to use to compare the my results, rather than just my
> opinion of the audio quality.
>
>
> Thanks,
>
> Adam Harding
>
> -----------------------------------------
> Email sent from www.virginmedia.com/email
> Virus-checked using McAfee(R) Software and scanned for spam
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
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From chris_christophersen at hotmail.com  Sat Mar 24 14:15:37 2007
From: chris_christophersen at hotmail.com (Chris Christophersen)
Date: Sat, 24 Mar 2007 14:15:37 -0400
Subject: [SIPForum-discussion] (no subject)
Message-ID: <BAY103-F14F32D527D6CD6A8736A5EFC690@phx.gbl>

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From deepanshu at huawei.com  Sun Mar 25 21:39:25 2007
From: deepanshu at huawei.com (Deepanshu)
Date: Mon, 26 Mar 2007 09:39:25 +0800
Subject: [SIPForum-discussion] How to implement a chat M2M
References: <d08f948c0703230359h55727018x62fe914d05c96de1@mail.gmail.com>
Message-ID: <00c301c76f47$96a7f7e0$8178a40a@china.huawei.com>

You can use the concept defined in draft-ietf-sipping-capacity-attribute-03.txt.

Or 

A Adhoc conference can be established by Client A and all other users can join in the conference. They can exchange the message using MSRP



HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC


  ----- Original Message ----- 
  From: Francesco Paolo Landolfo 
  To: discussion at sipforum.org 
  Sent: Friday, March 23, 2007 6:59 PM
  Subject: [SIPForum-discussion] How to implement a chat M2M


  Hi,
  I have implemented a chat P2P using SIP and the MESSAGE message. 
  Pratically, if Pippo want to send an instant message to Pluto, Pippo sends a MESSAGE to a Server, this Server convert the Pluto Sip Uri in Pluto Care of Address and send it to Pluto. This is in agreement with rfc3261. 

  Now I'd like to implement a chat M2M using SIP, but I have some doubt about flow and messages that I have to use.

  I think that there is one possible scene:

    a.. A client send a MESSAGE to a Server that dispatchs this message to all chat partecipant. 
  Now, how can I implement this feature? 
  Have someone some other idea?

  Thanks,
  Francesco

  -- 
  Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
  "Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto presto." (C'era una volta in America) 
  E adesso so cosa devo fare, devo continuare a respirare perch? domani il sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
  Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso) 


------------------------------------------------------------------------------


  _______________________________________________
  This is the SIP Forum discussion mailing list
  TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
  Post to the list at discussion at sipforum.org
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From Shanmukharao.Makkapati at airtel.in  Mon Mar 26 04:29:13 2007
From: Shanmukharao.Makkapati at airtel.in (Shanmukharao.Makkapati at airtel.in)
Date: Mon, 26 Mar 2007 13:59:13 +0530
Subject: [SIPForum-discussion] Interview Faqs
Message-ID: <OF05C5D884.B386EE92-ON652572AA.002E871B-652572AA.002E3F76@bharti.com>


Dear All,
           Can anyone send me the interview faqs on sip protocol...Please


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From umair3210 at yahoo.com  Mon Mar 26 07:18:06 2007
From: umair3210 at yahoo.com (Muhammad Umair)
Date: Mon, 26 Mar 2007 04:18:06 -0700 (PDT)
Subject: [SIPForum-discussion] softphone's help needed
Message-ID: <362122.81927.qm@web38714.mail.mud.yahoo.com>

hi all,
   
  i m umair , i m a student of final year computer science n IT. i m going to make my final project on soft phones. can any one tell me where to start from, practical implementations of SIP n some thing about practical developement environment of softphones.
   
  thank u

 
---------------------------------
8:00? 8:25? 8:40?  Find a flick in no time
 with theYahoo! Search movie showtime shortcut.
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From niklas.fondberg at tilgin.com  Mon Mar 26 15:30:24 2007
From: niklas.fondberg at tilgin.com (Niklas Fondberg)
Date: Mon, 26 Mar 2007 21:30:24 +0200
Subject: [SIPForum-discussion] simultaneous INVITEs
Message-ID: <1174937424.5519.12.camel@localhost.localdomain>

Hi, 
I new to this list but I hope that the list is what I'm after; an
implementation and design discussion list about SIP.
If my question is wrongly addressed, please forgive me and please point
me the right direction...

My question that I have searched all over for an answer to is quite
simple:

What is the correct behavior for a UA if a second INVITE arrives before
the first has been answered?

SIP Flow:

-------------> INVITE (1)
<------------- 100 Trying (1)
<------------- 180 Ringing (1)
-------------> INVITE (2)
... ???

Here the first (1) INVITE could have been answered by some other UA that
the INVITE might have been forked to and in this case session (2) should
start ringing. 


Niklas Fondberg




From adam.harding2 at ntlworld.com  Mon Mar 26 18:01:13 2007
From: adam.harding2 at ntlworld.com (Adam Harding)
Date: Mon, 26 Mar 2007 23:01:13 +0100
Subject: [SIPForum-discussion] R-Factor type equation to evaluate VOIP
	quality from Wireshark RTP stats
Message-ID: <20070326220113.LRUQ17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>

Hi,

I am interested in any free algorithms that can be used to give a value for the voice quality in a VOIP call based on basic statistics such as delay, packet loss and jitter.

I asked this question a few days and got a really useful document recommended to me which helps me understand how the R-Factor works but I can not get hold of the ITU-G values and my RTP results from Wireshark are probably to basic to calculate the R-Factor.

So just wondering if there is some sort of basic algorithm that I can enter my results from the Wireshark RTP stats and get some sort of value of voice quality that I can use to compare my results with each other. 

Thanks,

Adam Harding

-----------------------------------------
Email sent from www.virginmedia.com/email
Virus-checked using McAfee(R) Software and scanned for spam


From deepanshu at huawei.com  Mon Mar 26 21:50:17 2007
From: deepanshu at huawei.com (Deepanshu)
Date: Tue, 27 Mar 2007 09:50:17 +0800
Subject: [SIPForum-discussion] simultaneous INVITEs
References: <1174937424.5519.12.camel@localhost.localdomain>
Message-ID: <003801c77012$45a10050$8178a40a@china.huawei.com>

inline
----- Original Message -----
From: "Niklas Fondberg" <niklas.fondberg at tilgin.com>
To: <discussion at sipforum.org>
Sent: Tuesday, March 27, 2007 3:30 AM
Subject: [SIPForum-discussion] simultaneous INVITEs


> Hi,
> I new to this list but I hope that the list is what I'm after; an
> implementation and design discussion list about SIP.
> If my question is wrongly addressed, please forgive me and please point
> me the right direction...
>
> My question that I have searched all over for an answer to is quite
> simple:
>
> What is the correct behavior for a UA if a second INVITE arrives before
> the first has been answered?
>
> SIP Flow:
>
> -------------> INVITE (1)
> <------------- 100 Trying (1)
> <------------- 180 Ringing (1)
> -------------> INVITE (2)
> ... ???
>
> Here the first (1) INVITE could have been answered by some other UA that
> the INVITE might have been forked to and in this case session (2) should
> start ringing.

If the first (1) INVITE could have been answered by some other UA then the
originating UAC SHALL send a CANCEL request towards UAS instead of a INVITE
(2).

-------------> INVITE (1)
<------------- 100 Trying (1)
<------------- 180 Ringing (1)
--------------> CANCEL (1)
                        stop ringing
<------------- SIP 487 (1)


HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC

>
>
> Niklas Fondberg
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>




From belic_sonja at yahoo.com  Tue Mar 27 11:05:23 2007
From: belic_sonja at yahoo.com (Sonja Belic)
Date: Tue, 27 Mar 2007 08:05:23 -0700 (PDT)
Subject: [SIPForum-discussion] Authentication and authorization in SIP
Message-ID: <664094.34155.qm@web60623.mail.yahoo.com>

Hi,
 I have a question regarding authentication and authorization mechanism in SIP. For instance, if there are more then one applications running on the same SIP system, does every application authenticate itself or all of them use the same authentication parameters, defined for that SIP system?
 Thanks in advance.
 
 Best Regards,
 Sonja
 
---------------------------------
No need to miss a message. Get email on-the-go 
with Yahoo! Mail for Mobile. Get started.
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From qt.kiran at gmail.com  Tue Mar 27 12:25:47 2007
From: qt.kiran at gmail.com (kiran chakkilam)
Date: Tue, 27 Mar 2007 21:55:47 +0530
Subject: [SIPForum-discussion] content length
Message-ID: <e0646ce70703270925i183a21faocb49e8158212f9b2@mail.gmail.com>

Hi all,


How to calculate message body in sipp. Any open source tools are there to
calculate message body. please help me . It's very urgent for me.






Thanks in Advance

ch.kiran
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From adam.harding2 at ntlworld.com  Tue Mar 27 15:02:25 2007
From: adam.harding2 at ntlworld.com (Adam Harding)
Date: Tue, 27 Mar 2007 20:02:25 +0100
Subject: [SIPForum-discussion] RTP Delta and Difference : whats the
	difference?!
Message-ID: <20070327190225.BWMA17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>

Hi,

Regarding RTP analysis in Wireshark/Ethereal I am confused between the difference between "Delta" which I think is the delay between 2 consecutive packets and the "difference" value that is indicated on the graph given in ethereal. 

How is the difference value indicated on the graph calculated? Is there a forumla?

Thanks.

-----------------------------------------
Email sent from www.virginmedia.com/email
Virus-checked using McAfee(R) Software and scanned for spam


From wang.ran at byd.com.cn  Tue Mar 27 23:31:43 2007
From: wang.ran at byd.com.cn (wangran)
Date: Wed, 28 Mar 2007 11:31:43 +0800
Subject: [SIPForum-discussion]
	=?gb2312?B?tPC4tDogZGlzY3Vzc2lvbiBEaWdlc3QsIFZvbCAyMCwgSXNzdWUgMzg=?=
In-Reply-To: <mailman.1.1175011201.26118.discussion@sipforum.org>
Message-ID: <OF0ECD4452.9BFB2601-ON482572AC.00137279@byd.com.cn>

Hi..
         We have some question in RFC3261, May I have you some minute to
explain it?
         Alice and bob??s call flaw in chapter 4 figure.1 is like this:

         SIP Flow:

-------------> INVITE (F1)\

-------------> INVITE (F2)

<------------- 100 Trying (F3)

         

         But in chapter 24.2

         F2 and F3 exchange there sequence

-------------> INVITE (F1)\

<------------- 100 Trying (F2)

-------------> INVITE (F3)

 

Does this small difference cause problems?

your comment will be highly appreciated!

 


Best of Regards,
 
wangran
 
 ***********************************************************************
BYD TECHFAITH??COMPANY??LIMITED(BTC)
 
Address:3/F,M8 West,NO.1 Jiu Xian Qiao Dong Road,
        Chao Yang District,Beijing,China  100016
PostCode:10016
Mobile: +86-13810362150
Tel: +86-10-58291226
Mail: wang.ran at byd.com.cn
 ***********************************************************************
 
 
-----????????-----
??????: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.
org] ???? discussion-request at sipforum.org
????????: 2007??3??28?? 0:00
??????: discussion at sipforum.org
????: discussion Digest, Vol 20, Issue 38

Send discussion mailing list submissions to
	discussion at sipforum.org

To subscribe or unsubscribe via the World Wide Web, visit
	http://sipforum.org/mailman/listinfo/discussion
or, via email, send a message with subject or body 'help' to
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	discussion-owner at sipforum.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of discussion digest..."


Today's Topics:

   1. simultaneous INVITEs (Niklas Fondberg)
   2. R-Factor type equation to evaluate VOIP	quality from
      Wireshark RTP stats (Adam Harding)
   3. Re: simultaneous INVITEs (Deepanshu)
   4. Authentication and authorization in SIP (Sonja Belic)


----------------------------------------------------------------------

Message: 1
Date: Mon, 26 Mar 2007 21:30:24 +0200
From: Niklas Fondberg <niklas.fondberg at tilgin.com>
Subject: [SIPForum-discussion] simultaneous INVITEs
To: discussion at sipforum.org
Message-ID: <1174937424.5519.12.camel at localhost.localdomain>
Content-Type: text/plain

Hi, 
I new to this list but I hope that the list is what I'm after; an
implementation and design discussion list about SIP.
If my question is wrongly addressed, please forgive me and please point
me the right direction...

My question that I have searched all over for an answer to is quite
simple:

What is the correct behavior for a UA if a second INVITE arrives before
the first has been answered?

SIP Flow:

-------------> INVITE (1)
<------------- 100 Trying (1)
<------------- 180 Ringing (1)
-------------> INVITE (2)
... ???

Here the first (1) INVITE could have been answered by some other UA that
the INVITE might have been forked to and in this case session (2) should
start ringing. 


Niklas Fondberg





------------------------------

Message: 2
Date: Mon, 26 Mar 2007 23:01:13 +0100
From: Adam Harding <adam.harding2 at ntlworld.com>
Subject: [SIPForum-discussion] R-Factor type equation to evaluate VOIP
	quality from Wireshark RTP stats
To: "discussion at sipforum.org" <discussion at sipforum.org>
Message-ID:
	
<20070326220113.LRUQ17393.aamtaout02-winn.ispmail.ntl.com at smtp.ntlworld.com>
	
Content-Type: text/plain; charset=ISO-8859-1

Hi,

I am interested in any free algorithms that can be used to give a value for
the voice quality in a VOIP call based on basic statistics such as delay,
packet loss and jitter.

I asked this question a few days and got a really useful document
recommended to me which helps me understand how the R-Factor works but I can
not get hold of the ITU-G values and my RTP results from Wireshark are
probably to basic to calculate the R-Factor.

So just wondering if there is some sort of basic algorithm that I can enter
my results from the Wireshark RTP stats and get some sort of value of voice
quality that I can use to compare my results with each other. 

Thanks,

Adam Harding

-----------------------------------------
Email sent from www.virginmedia.com/email
Virus-checked using McAfee(R) Software and scanned for spam



------------------------------

Message: 3
Date: Tue, 27 Mar 2007 09:50:17 +0800
From: Deepanshu <deepanshu at huawei.com>
Subject: Re: [SIPForum-discussion] simultaneous INVITEs
To: Niklas Fondberg <niklas.fondberg at tilgin.com>
Cc: discussion at sipforum.org
Message-ID: <003801c77012$45a10050$8178a40a at china.huawei.com>
Content-Type: text/plain; charset=iso-8859-1

inline
----- Original Message -----
From: "Niklas Fondberg" <niklas.fondberg at tilgin.com>
To: <discussion at sipforum.org>
Sent: Tuesday, March 27, 2007 3:30 AM
Subject: [SIPForum-discussion] simultaneous INVITEs


> Hi,
> I new to this list but I hope that the list is what I'm after; an
> implementation and design discussion list about SIP.
> If my question is wrongly addressed, please forgive me and please point
> me the right direction...
>
> My question that I have searched all over for an answer to is quite
> simple:
>
> What is the correct behavior for a UA if a second INVITE arrives before
> the first has been answered?
>
> SIP Flow:
>
> -------------> INVITE (1)
> <------------- 100 Trying (1)
> <------------- 180 Ringing (1)
> -------------> INVITE (2)
> ... ???
>
> Here the first (1) INVITE could have been answered by some other UA that
> the INVITE might have been forked to and in this case session (2) should
> start ringing.

If the first (1) INVITE could have been answered by some other UA then the
originating UAC SHALL send a CANCEL request towards UAS instead of a INVITE
(2).

-------------> INVITE (1)
<------------- 100 Trying (1)
<------------- 180 Ringing (1)
--------------> CANCEL (1)
                        stop ringing
<------------- SIP 487 (1)


HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC

>
>
> Niklas Fondberg
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>





------------------------------

Message: 4
Date: Tue, 27 Mar 2007 08:05:23 -0700 (PDT)
From: Sonja Belic <belic_sonja at yahoo.com>
Subject: [SIPForum-discussion] Authentication and authorization in SIP
To: discussion at sipforum.org
Message-ID: <664094.34155.qm at web60623.mail.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi,
 I have a question regarding authentication and authorization mechanism in
SIP. For instance, if there are more then one applications running on the
same SIP system, does every application authenticate itself or all of them
use the same authentication parameters, defined for that SIP system?
 Thanks in advance.
 
 Best Regards,
 Sonja
 
---------------------------------
No need to miss a message. Get email on-the-go 
with Yahoo! Mail for Mobile. Get started.
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From pallavim35 at gmail.com  Wed Mar 28 00:22:26 2007
From: pallavim35 at gmail.com (aditi g)
Date: Wed, 28 Mar 2007 09:52:26 +0530
Subject: [SIPForum-discussion] Why Ack is different transaction?
Message-ID: <63af059d0703272122l5134afpa51426b1551dd36@mail.gmail.com>

Hi,

I want to know why ACK is considered different transaction from Invite
transaction .

regs
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From skp10_9559 at yahoo.co.in  Wed Mar 28 00:29:13 2007
From: skp10_9559 at yahoo.co.in (santosh patra)
Date: Wed, 28 Mar 2007 09:59:13 +0530 (IST)
Subject: [SIPForum-discussion] pls help me in this basic scenario
Message-ID: <318897.10226.qm@web8809.mail.in.yahoo.com>

Hello Kiran

As far as my knowledge says SIPP does not support any header except SDP headers in shortcode format,it can be if you are able to change the source code of SIPp and might be calleg is hanging that's why your call is not getting terminated.

Regards
Santosh Patra


----- Original Message ----
From: kiran chakkilam <qt.kiran at gmail.com>
To: discussion at sipforum.org
Sent: Friday, 23 March, 2007 4:49:10 PM
Subject: [SIPForum-discussion] pls help me in this basic scenario


Hi everybody,
 
 
I am currently working on the sipp1.1 . i want to test proxy server(System Under test). so i am simulating UAC and UAS using sipp. i will 
 
send invite messge from UAC ,it will hit proxy(SUT).Proxy responds with 100 trying response back to UAC with all headers in short format 
 
then my UAC is terminated.It's showing an error  No valid call-id. so I have some doubts in sipp.
 
1. whether sipp accepts headers in shorcode format.
 
2. why it's terminating after receiving the 100 trying response.
 
 
so kindly help me in this scenario,it's very urgent for me
 
 
Thanks & Regards
 
ch.kiran
 
 
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From skp10_9559 at yahoo.co.in  Wed Mar 28 00:35:51 2007
From: skp10_9559 at yahoo.co.in (santosh patra)
Date: Wed, 28 Mar 2007 10:05:51 +0530 (IST)
Subject: [SIPForum-discussion] simultaneous INVITEs
Message-ID: <657666.6270.qm@web8807.mail.in.yahoo.com>

Hello Niklas

In this scenario, lets take a pratical example that
1. A calls B
2. B is ringing
3. In the mean time if C calls B then though A and B's call has not established, C should get busy response or if the A and B's call is established and B has call waiting activated then C han hear the Ringing of B user and B will get the indication that somebody is  calliing.

Santosh


----- Original Message ----
From: Niklas Fondberg <niklas.fondberg at tilgin.com>
To: discussion at sipforum.org
Sent: Tuesday, 27 March, 2007 1:00:24 AM
Subject: [SIPForum-discussion] simultaneous INVITEs


Hi, 
I new to this list but I hope that the list is what I'm after; an
implementation and design discussion list about SIP.
If my question is wrongly addressed, please forgive me and please point
me the right direction...

My question that I have searched all over for an answer to is quite
simple:

What is the correct behavior for a UA if a second INVITE arrives before
the first has been answered?

SIP Flow:

-------------> INVITE (1)
<------------- 100 Trying (1)
<------------- 180 Ringing (1)
-------------> INVITE (2)
... ???

Here the first (1) INVITE could have been answered by some other UA that
the INVITE might have been forked to and in this case session (2) should
start ringing. 


Niklas Fondberg



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From deepanshu at huawei.com  Wed Mar 28 01:58:41 2007
From: deepanshu at huawei.com (Deepanshu)
Date: Wed, 28 Mar 2007 13:58:41 +0800
Subject: [SIPForum-discussion]
	=?gb2312?B?UmU6IFtTSVBGb3J1bS1kaXNjdXNzaW9uXbTwuLQ6IGRpc2N1c3Npbw==?=	=?gb2312?B?biBEaWdlc3QsIFZvbCAyMCwgSXNzdWUgMzg=?=
References: <OF0ECD4452.9BFB2601-ON482572AC.00137279@byd.com.cn>
Message-ID: <00b601c770fe$23466ad0$8178a40a@china.huawei.com>

Dear Wang

I don't think this create any kind of problems. 100 trying is hop-by-hop,
the proxy can perform it simultaneously with other request (INVITE P1--->P2
in your case)

HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC

----- Original Message -----
From: "wangran" <wang.ran at byd.com.cn>
To: <discussion at sipforum.org>
Sent: Wednesday, March 28, 2007 11:31 AM
Subject: [SIPForum-discussion]????: discussion Digest, Vol 20, Issue 38


> Hi..
>          We have some question in RFC3261, May I have you some minute to
> explain it?
>          Alice and bob??s call flaw in chapter 4 figure.1 is like this:
>
>          SIP Flow:
>
> -------------> INVITE (F1)\
>
> -------------> INVITE (F2)
>
> <------------- 100 Trying (F3)
>
>
>
>          But in chapter 24.2
>
>          F2 and F3 exchange there sequence
>
> -------------> INVITE (F1)\
>
> <------------- 100 Trying (F2)
>
> -------------> INVITE (F3)
>
>
>
> Does this small difference cause problems?
>
> your comment will be highly appreciated!
>
>
>
>
> Best of Regards,
>
> wangran
>
>  ***********************************************************************
> BYD TECHFAITH??COMPANY??LIMITED(BTC)
>
> Address:3/F,M8 West,NO.1 Jiu Xian Qiao Dong Road,
>         Chao Yang District,Beijing,China  100016
> PostCode:10016
> Mobile: +86-13810362150
> Tel: +86-10-58291226
> Mail: wang.ran at byd.com.cn
>  ***********************************************************************
>
>
> -----????????-----
> ??????: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.
> org] ???? discussion-request at sipforum.org
> ????????: 2007??3??28?? 0:00
> ??????: discussion at sipforum.org
> ????: discussion Digest, Vol 20, Issue 38
>
> Send discussion mailing list submissions to
> discussion at sipforum.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://sipforum.org/mailman/listinfo/discussion
> or, via email, send a message with subject or body 'help' to
> discussion-request at sipforum.org
>
> You can reach the person managing the list at
> discussion-owner at sipforum.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of discussion digest..."
>
>
> Today's Topics:
>
>    1. simultaneous INVITEs (Niklas Fondberg)
>    2. R-Factor type equation to evaluate VOIP quality from
>       Wireshark RTP stats (Adam Harding)
>    3. Re: simultaneous INVITEs (Deepanshu)
>    4. Authentication and authorization in SIP (Sonja Belic)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 26 Mar 2007 21:30:24 +0200
> From: Niklas Fondberg <niklas.fondberg at tilgin.com>
> Subject: [SIPForum-discussion] simultaneous INVITEs
> To: discussion at sipforum.org
> Message-ID: <1174937424.5519.12.camel at localhost.localdomain>
> Content-Type: text/plain
>
> Hi,
> I new to this list but I hope that the list is what I'm after; an
> implementation and design discussion list about SIP.
> If my question is wrongly addressed, please forgive me and please point
> me the right direction...
>
> My question that I have searched all over for an answer to is quite
> simple:
>
> What is the correct behavior for a UA if a second INVITE arrives before
> the first has been answered?
>
> SIP Flow:
>
> -------------> INVITE (1)
> <------------- 100 Trying (1)
> <------------- 180 Ringing (1)
> -------------> INVITE (2)
> ... ???
>
> Here the first (1) INVITE could have been answered by some other UA that
> the INVITE might have been forked to and in this case session (2) should
> start ringing.
>
>
> Niklas Fondberg
>
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Mon, 26 Mar 2007 23:01:13 +0100
> From: Adam Harding <adam.harding2 at ntlworld.com>
> Subject: [SIPForum-discussion] R-Factor type equation to evaluate VOIP
> quality from Wireshark RTP stats
> To: "discussion at sipforum.org" <discussion at sipforum.org>
> Message-ID:
>
>
<20070326220113.LRUQ17393.aamtaout02-winn.ispmail.ntl.com at smtp.ntlworld.com>
>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hi,
>
> I am interested in any free algorithms that can be used to give a value
for
> the voice quality in a VOIP call based on basic statistics such as delay,
> packet loss and jitter.
>
> I asked this question a few days and got a really useful document
> recommended to me which helps me understand how the R-Factor works but I
can
> not get hold of the ITU-G values and my RTP results from Wireshark are
> probably to basic to calculate the R-Factor.
>
> So just wondering if there is some sort of basic algorithm that I can
enter
> my results from the Wireshark RTP stats and get some sort of value of
voice
> quality that I can use to compare my results with each other.
>
> Thanks,
>
> Adam Harding
>
> -----------------------------------------
> Email sent from www.virginmedia.com/email
> Virus-checked using McAfee(R) Software and scanned for spam
>
>
>
> ------------------------------
>
> Message: 3
> Date: Tue, 27 Mar 2007 09:50:17 +0800
> From: Deepanshu <deepanshu at huawei.com>
> Subject: Re: [SIPForum-discussion] simultaneous INVITEs
> To: Niklas Fondberg <niklas.fondberg at tilgin.com>
> Cc: discussion at sipforum.org
> Message-ID: <003801c77012$45a10050$8178a40a at china.huawei.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> inline
> ----- Original Message -----
> From: "Niklas Fondberg" <niklas.fondberg at tilgin.com>
> To: <discussion at sipforum.org>
> Sent: Tuesday, March 27, 2007 3:30 AM
> Subject: [SIPForum-discussion] simultaneous INVITEs
>
>
> > Hi,
> > I new to this list but I hope that the list is what I'm after; an
> > implementation and design discussion list about SIP.
> > If my question is wrongly addressed, please forgive me and please point
> > me the right direction...
> >
> > My question that I have searched all over for an answer to is quite
> > simple:
> >
> > What is the correct behavior for a UA if a second INVITE arrives before
> > the first has been answered?
> >
> > SIP Flow:
> >
> > -------------> INVITE (1)
> > <------------- 100 Trying (1)
> > <------------- 180 Ringing (1)
> > -------------> INVITE (2)
> > ... ???
> >
> > Here the first (1) INVITE could have been answered by some other UA that
> > the INVITE might have been forked to and in this case session (2) should
> > start ringing.
>
> If the first (1) INVITE could have been answered by some other UA then the
> originating UAC SHALL send a CANCEL request towards UAS instead of a
INVITE
> (2).
>
> -------------> INVITE (1)
> <------------- 100 Trying (1)
> <------------- 180 Ringing (1)
> --------------> CANCEL (1)
>                         stop ringing
> <------------- SIP 487 (1)
>
>
> HTH
>
> Deepanshu Gautam
> R&D Engineer
> Huawei Technologies Co. Ltd.
> Nanjing, PRC
>
> >
> >
> > Niklas Fondberg
> >
> >
> >
> > _______________________________________________
> > This is the SIP Forum discussion mailing list
> > TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> > Post to the list at discussion at sipforum.org
> >
>
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 27 Mar 2007 08:05:23 -0700 (PDT)
> From: Sonja Belic <belic_sonja at yahoo.com>
> Subject: [SIPForum-discussion] Authentication and authorization in SIP
> To: discussion at sipforum.org
> Message-ID: <664094.34155.qm at web60623.mail.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>  I have a question regarding authentication and authorization mechanism in
> SIP. For instance, if there are more then one applications running on the
> same SIP system, does every application authenticate itself or all of them
> use the same authentication parameters, defined for that SIP system?
>  Thanks in advance.
>
>  Best Regards,
>  Sonja
>
> ---------------------------------
> No need to miss a message. Get email on-the-go
> with Yahoo! Mail for Mobile. Get started.
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> An HTML attachment was scrubbed...
> URL:
>
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> hment-0001.html
>
> ------------------------------
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.
> org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
> End of discussion Digest, Vol 20, Issue 38
> ******************************************
>
>
> Powered by BYD Security Gateway.
>
>
>
> Powered by BYD Security Gateway.
>
>
>


----------------------------------------------------------------------------
----


> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
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>



From resalath.ahamed at gmail.com  Wed Mar 28 02:23:21 2007
From: resalath.ahamed at gmail.com (resalath ahamed)
Date: Wed, 28 Mar 2007 11:53:21 +0530
Subject: [SIPForum-discussion]
	=?GB2312?B?UmU6IFtTSVBGb3J1bS1kaXNjdXNzaW9uXSBSZTogW1NJUEZvcnVtLWRpc2N1?=
	=?GB2312?B?c3Npb25dtPC4tDogZGlzY3Vzc2lvbiBEaWdlc3QsIFZvbCAyMCwgSXNzdWUgMzg=?=
In-Reply-To: <00b601c770fe$23466ad0$8178a40a@china.huawei.com>
References: <OF0ECD4452.9BFB2601-ON482572AC.00137279@byd.com.cn>
	<00b601c770fe$23466ad0$8178a40a@china.huawei.com>
Message-ID: <3ec935670703272323pd28519cr9efcfe802c277034@mail.gmail.com>

wangran,

Your question can be solved by understanding "STATEFULL PROXY" and
"STATELESS PROXY".

Below are two scenarios:

[1] Call involves stateless proxy - Invite will be forwarded by a stateless
proxy without returning 100 trying to the originator. So in this case 100
trying should come from terminator or from other statefull proxy in the
network. The stateless proxy will forward the 100 trying to UAC. A stateless
proxy does not maintain the call state so it can not send 100 trying by its
own. It will only forward the 100 trying.

[2] Call involves statefull proxy - In this case a statefull proxy will
return 100 trying to the UAC before it forwards the request to the next SIP
element in the network. A statefull proxy maintains the call state and has
all the records of the call that is being processed. So it can trigger 100
trying to the originator.

Hope this solves your issue.

Thanks and Regards,
Resalath Ahamed.



On 3/28/07, Deepanshu <deepanshu at huawei.com> wrote:
>
> Dear Wang
>
> I don't think this create any kind of problems. 100 trying is hop-by-hop,
> the proxy can perform it simultaneously with other request (INVITE
> P1--->P2
> in your case)
>
> HTH
>
> Deepanshu Gautam
> R&D Engineer
> Huawei Technologies Co. Ltd.
> Nanjing, PRC
>
> ----- Original Message -----
> From: "wangran" <wang.ran at byd.com.cn>
> To: <discussion at sipforum.org>
> Sent: Wednesday, March 28, 2007 11:31 AM
> Subject: [SIPForum-discussion]????: discussion Digest, Vol 20, Issue 38
>
>
> > Hi..
> >          We have some question in RFC3261, May I have you some minute to
> > explain it?
> >          Alice and bob's call flaw in chapter 4 figure.1 is like this:
> >
> >          SIP Flow:
> >
> > -------------> INVITE (F1)\
> >
> > -------------> INVITE (F2)
> >
> > <------------- 100 Trying (F3)
> >
> >
> >
> >          But in chapter 24.2
> >
> >          F2 and F3 exchange there sequence
> >
> > -------------> INVITE (F1)\
> >
> > <------------- 100 Trying (F2)
> >
> > -------------> INVITE (F3)
> >
> >
> >
> > Does this small difference cause problems?
> >
> > your comment will be highly appreciated!
> >
> >
> >
> >
> > Best of Regards,
> >
> > wangran
> >
> >  ***********************************************************************
> > BYD TECHFAITH COMPANY LIMITED(BTC)
> >
> > Address:3/F,M8 West,NO.1 Jiu Xian Qiao Dong Road,
> >         Chao Yang District,Beijing,China  100016
> > PostCode:10016
> > Mobile: +86-13810362150
> > Tel: +86-10-58291226
> > Mail: wang.ran at byd.com.cn
> >  ***********************************************************************
> >
> >
> > -----????????-----
> > ??????: discussion-bounces at sipforum.org
> [mailto:discussion-bounces at sipforum.
> > org] ???? discussion-request at sipforum.org
> > ????????: 2007??3??28?? 0:00
> > ??????: discussion at sipforum.org
> > ????: discussion Digest, Vol 20, Issue 38
> >
> > Send discussion mailing list submissions to
> > discussion at sipforum.org
> >
> > To subscribe or unsubscribe via the World Wide Web, visit
> > http://sipforum.org/mailman/listinfo/discussion
> > or, via email, send a message with subject or body 'help' to
> > discussion-request at sipforum.org
> >
> > You can reach the person managing the list at
> > discussion-owner at sipforum.org
> >
> > When replying, please edit your Subject line so it is more specific
> > than "Re: Contents of discussion digest..."
> >
> >
> > Today's Topics:
> >
> >    1. simultaneous INVITEs (Niklas Fondberg)
> >    2. R-Factor type equation to evaluate VOIP quality from
> >       Wireshark RTP stats (Adam Harding)
> >    3. Re: simultaneous INVITEs (Deepanshu)
> >    4. Authentication and authorization in SIP (Sonja Belic)
> >
> >
> > ----------------------------------------------------------------------
> >
> > Message: 1
> > Date: Mon, 26 Mar 2007 21:30:24 +0200
> > From: Niklas Fondberg <niklas.fondberg at tilgin.com>
> > Subject: [SIPForum-discussion] simultaneous INVITEs
> > To: discussion at sipforum.org
> > Message-ID: <1174937424.5519.12.camel at localhost.localdomain>
> > Content-Type: text/plain
> >
> > Hi,
> > I new to this list but I hope that the list is what I'm after; an
> > implementation and design discussion list about SIP.
> > If my question is wrongly addressed, please forgive me and please point
> > me the right direction...
> >
> > My question that I have searched all over for an answer to is quite
> > simple:
> >
> > What is the correct behavior for a UA if a second INVITE arrives before
> > the first has been answered?
> >
> > SIP Flow:
> >
> > -------------> INVITE (1)
> > <------------- 100 Trying (1)
> > <------------- 180 Ringing (1)
> > -------------> INVITE (2)
> > ... ???
> >
> > Here the first (1) INVITE could have been answered by some other UA that
> > the INVITE might have been forked to and in this case session (2) should
> > start ringing.
> >
> >
> > Niklas Fondberg
> >
> >
> >
> >
> >
> > ------------------------------
> >
> > Message: 2
> > Date: Mon, 26 Mar 2007 23:01:13 +0100
> > From: Adam Harding <adam.harding2 at ntlworld.com>
> > Subject: [SIPForum-discussion] R-Factor type equation to evaluate VOIP
> > quality from Wireshark RTP stats
> > To: "discussion at sipforum.org" <discussion at sipforum.org>
> > Message-ID:
> >
> >
> <
> 20070326220113.LRUQ17393.aamtaout02-winn.ispmail.ntl.com at smtp.ntlworld.com
> >
> >
> > Content-Type: text/plain; charset=ISO-8859-1
> >
> > Hi,
> >
> > I am interested in any free algorithms that can be used to give a value
> for
> > the voice quality in a VOIP call based on basic statistics such as
> delay,
> > packet loss and jitter.
> >
> > I asked this question a few days and got a really useful document
> > recommended to me which helps me understand how the R-Factor works but I
> can
> > not get hold of the ITU-G values and my RTP results from Wireshark are
> > probably to basic to calculate the R-Factor.
> >
> > So just wondering if there is some sort of basic algorithm that I can
> enter
> > my results from the Wireshark RTP stats and get some sort of value of
> voice
> > quality that I can use to compare my results with each other.
> >
> > Thanks,
> >
> > Adam Harding
> >
> > -----------------------------------------
> > Email sent from www.virginmedia.com/email
> > Virus-checked using McAfee(R) Software and scanned for spam
> >
> >
> >
> > ------------------------------
> >
> > Message: 3
> > Date: Tue, 27 Mar 2007 09:50:17 +0800
> > From: Deepanshu <deepanshu at huawei.com>
> > Subject: Re: [SIPForum-discussion] simultaneous INVITEs
> > To: Niklas Fondberg <niklas.fondberg at tilgin.com>
> > Cc: discussion at sipforum.org
> > Message-ID: <003801c77012$45a10050$8178a40a at china.huawei.com>
> > Content-Type: text/plain; charset=iso-8859-1
> >
> > inline
> > ----- Original Message -----
> > From: "Niklas Fondberg" <niklas.fondberg at tilgin.com>
> > To: <discussion at sipforum.org>
> > Sent: Tuesday, March 27, 2007 3:30 AM
> > Subject: [SIPForum-discussion] simultaneous INVITEs
> >
> >
> > > Hi,
> > > I new to this list but I hope that the list is what I'm after; an
> > > implementation and design discussion list about SIP.
> > > If my question is wrongly addressed, please forgive me and please
> point
> > > me the right direction...
> > >
> > > My question that I have searched all over for an answer to is quite
> > > simple:
> > >
> > > What is the correct behavior for a UA if a second INVITE arrives
> before
> > > the first has been answered?
> > >
> > > SIP Flow:
> > >
> > > -------------> INVITE (1)
> > > <------------- 100 Trying (1)
> > > <------------- 180 Ringing (1)
> > > -------------> INVITE (2)
> > > ... ???
> > >
> > > Here the first (1) INVITE could have been answered by some other UA
> that
> > > the INVITE might have been forked to and in this case session (2)
> should
> > > start ringing.
> >
> > If the first (1) INVITE could have been answered by some other UA then
> the
> > originating UAC SHALL send a CANCEL request towards UAS instead of a
> INVITE
> > (2).
> >
> > -------------> INVITE (1)
> > <------------- 100 Trying (1)
> > <------------- 180 Ringing (1)
> > --------------> CANCEL (1)
> >                         stop ringing
> > <------------- SIP 487 (1)
> >
> >
> > HTH
> >
> > Deepanshu Gautam
> > R&D Engineer
> > Huawei Technologies Co. Ltd.
> > Nanjing, PRC
> >
> > >
> > >
> > > Niklas Fondberg
> > >
> > >
> > >
> > > _______________________________________________
> > > This is the SIP Forum discussion mailing list
> > > TO UNSUBSCRIBE, or edit your delivery options, please visit
> > http://sipforum.org/mailman/listinfo/discussion
> > > Post to the list at discussion at sipforum.org
> > >
> >
> >
> >
> >
> >
> > ------------------------------
> >
> > Message: 4
> > Date: Tue, 27 Mar 2007 08:05:23 -0700 (PDT)
> > From: Sonja Belic <belic_sonja at yahoo.com>
> > Subject: [SIPForum-discussion] Authentication and authorization in SIP
> > To: discussion at sipforum.org
> > Message-ID: <664094.34155.qm at web60623.mail.yahoo.com>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Hi,
> >  I have a question regarding authentication and authorization mechanism
> in
> > SIP. For instance, if there are more then one applications running on
> the
> > same SIP system, does every application authenticate itself or all of
> them
> > use the same authentication parameters, defined for that SIP system?
> >  Thanks in advance.
> >
> >  Best Regards,
> >  Sonja
> >
> > ---------------------------------
> > No need to miss a message. Get email on-the-go
> > with Yahoo! Mail for Mobile. Get started.
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> >
> > ------------------------------
> >
> > _______________________________________________
> > This is the SIP Forum discussion mailing list
> > TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.
> > org/mailman/listinfo/discussion
> > Post to the list at discussion at sipforum.org
> >
> >
> > End of discussion Digest, Vol 20, Issue 38
> > ******************************************
> >
> >
> > Powered by BYD Security Gateway.
> >
> >
> >
> > Powered by BYD Security Gateway.
> >
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> >
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> > TO UNSUBSCRIBE, or edit your delivery options, please visit
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From avishekchowdhury at tataelxsi.co.in  Wed Mar 28 02:37:35 2007
From: avishekchowdhury at tataelxsi.co.in (Avishek Chowdhury)
Date: Wed, 28 Mar 2007 12:07:35 +0530
Subject: [SIPForum-discussion] Why Ack is different transaction?
References: <63af059d0703272122l5134afpa51426b1551dd36@mail.gmail.com>
Message-ID: <0fd201c77103$92b192a0$6c19320a@telxsi.com>

Hi Aditi,

When the UAC receives 200 OK, the client transaction is destroyed at TL.
This is done because, the TL is unaware of the above core. The above core can be a Proxy or an UAC core.
In case of proxy, the 200 OK is forwarded whereas in case of UAC, an ACK is generated. Now this ACK has to create a new transaction (transaction created by INVITE had been destroyed)
at TL for its transmission, hence the ACK for 200 OK is outside the INVITE transaction.

For other non-2xx final responses, the client transaction at TL is not destroyed and the ACK is generated by TL. Hence in this case, the ACK is within the transaction.

Regards,
Avishek

  ----- Original Message ----- 
  From: aditi g 
  To: discussion at sipforum.org 
  Sent: Wednesday, March 28, 2007 9:52 AM
  Subject: [SIPForum-discussion] Why Ack is different transaction?


  Hi,

  I want to know why ACK is considered different transaction from Invite transaction .

  regs


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From sakcahalit at hotmail.com  Wed Mar 28 03:20:51 2007
From: sakcahalit at hotmail.com (Halit Sakca)
Date: Wed, 28 Mar 2007 10:20:51 +0300
Subject: [SIPForum-discussion] Authentication and authorization in SIP
Message-ID: <BAY126-W670CB6023CC4F864D6932CC6D0@phx.gbl>

Hi Sonja,Yes multiple applications can use same AA parameters, if you ask how, let me say that these parameters can be defined on a instance of DB server, so during the call handling, AA mechanism can  respond  to  multiple application.Thats my comment but I dont know that we are talking about same subject :)HalitDate: Tue, 27 Mar 2007 08:05:23 -0700From: belic_sonja at yahoo.comTo: discussion at sipforum.orgSubject: [SIPForum-discussion] Authentication and authorization in SIPHi, I have a question regarding authentication and authorization mechanism in SIP. For instance, if there are more then one applications running on the same SIP system, does every application authenticate itself or all of them use the same authentication parameters, defined for that SIP system? Thanks in advance.  Best Regards, Sonja 

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From gowshan at sltnet.lk  Wed Mar 28 02:58:28 2007
From: gowshan at sltnet.lk (shankar)
Date: Wed, 28 Mar 2007 12:58:28 +0600
Subject: [SIPForum-discussion] Web based sip client
In-Reply-To: <00b601c770fe$23466ad0$8178a40a@china.huawei.com>
Message-ID: <0JFL006DOPYNQWB0@pop3.sltnet.lk>


Hi 

Please anyone give me free web based sip client to put in my web page to
communicate with my sip server

Please provide program to put in my web server 


Thanks

Regards
Shankar


From alexzhang at gdnt.com.cn  Wed Mar 28 03:48:32 2007
From: alexzhang at gdnt.com.cn (alexzhang at gdnt.com.cn)
Date: Wed, 28 Mar 2007 15:48:32 +0800
Subject: [SIPForum-discussion]
	=?gb2312?B?UkU6IFtTSVBGb3J1bS1kaXNjdXNzaW9uXVJlOiBbU0lQRm9ydW0tZA==?=	=?gb2312?B?aXNjdXNzaW9uXbTwuLQ6IGRpc2N1c3NpbwluIERpZ2VzdCwgVm9sIDIwLCA=?=	=?gb2312?B?SXNzdWUgMzg=?=
In-Reply-To: <00b601c770fe$23466ad0$8178a40a@china.huawei.com>
References: <OF0ECD4452.9BFB2601-ON482572AC.00137279@byd.com.cn>
	<00b601c770fe$23466ad0$8178a40a@china.huawei.com>
Message-ID: <8E523FC208B8174790E69947E307914701770975@rnd-ex01.rnd.gdnt.local>

Anybody in this list are involved in the development of the SIP-I (SIP w/ecanpsulated ISUP) ?


- A L E X

-----Original Message-----
From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Deepanshu
Sent: Wednesday, March 28, 2007 1:59 PM
To: wangran; discussion at sipforum.org
Subject: [SIPForum-discussion]Re: [SIPForum-discussion]????: discussio n Digest, Vol 20, Issue 38

Dear Wang

I don't think this create any kind of problems. 100 trying is hop-by-hop, the proxy can perform it simultaneously with other request (INVITE P1--->P2 in your case)

HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC

----- Original Message -----
From: "wangran" <wang.ran at byd.com.cn>
To: <discussion at sipforum.org>
Sent: Wednesday, March 28, 2007 11:31 AM
Subject: [SIPForum-discussion]????: discussion Digest, Vol 20, Issue 38


> Hi..
>          We have some question in RFC3261, May I have you some minute 
> to explain it?
>          Alice and bob??s call flaw in chapter 4 figure.1 is like this:
>
>          SIP Flow:
>
> -------------> INVITE (F1)\
>
> -------------> INVITE (F2)
>
> <------------- 100 Trying (F3)
>
>
>
>          But in chapter 24.2
>
>          F2 and F3 exchange there sequence
>
> -------------> INVITE (F1)\
>
> <------------- 100 Trying (F2)
>
> -------------> INVITE (F3)
>
>
>
> Does this small difference cause problems?
>
> your comment will be highly appreciated!
>
>
>
>
> Best of Regards,
>
> wangran
>
>  
> **********************************************************************
> *
> BYD TECHFAITH??COMPANY??LIMITED(BTC)
>
> Address:3/F,M8 West,NO.1 Jiu Xian Qiao Dong Road,
>         Chao Yang District,Beijing,China  100016
> PostCode:10016
> Mobile: +86-13810362150
> Tel: +86-10-58291226
> Mail: wang.ran at byd.com.cn
>  
> **********************************************************************
> *
>
>
> -----????????-----
> ??????: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.
> org] ???? discussion-request at sipforum.org
> ????????: 2007??3??28?? 0:00
> ??????: discussion at sipforum.org
> ????: discussion Digest, Vol 20, Issue 38
>
> Send discussion mailing list submissions to discussion at sipforum.org
>
> To subscribe or unsubscribe via the World Wide Web, visit 
> http://sipforum.org/mailman/listinfo/discussion
> or, via email, send a message with subject or body 'help' to 
> discussion-request at sipforum.org
>
> You can reach the person managing the list at 
> discussion-owner at sipforum.org
>
> When replying, please edit your Subject line so it is more specific 
> than "Re: Contents of discussion digest..."
>
>
> Today's Topics:
>
>    1. simultaneous INVITEs (Niklas Fondberg)
>    2. R-Factor type equation to evaluate VOIP quality from
>       Wireshark RTP stats (Adam Harding)
>    3. Re: simultaneous INVITEs (Deepanshu)
>    4. Authentication and authorization in SIP (Sonja Belic)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 26 Mar 2007 21:30:24 +0200
> From: Niklas Fondberg <niklas.fondberg at tilgin.com>
> Subject: [SIPForum-discussion] simultaneous INVITEs
> To: discussion at sipforum.org
> Message-ID: <1174937424.5519.12.camel at localhost.localdomain>
> Content-Type: text/plain
>
> Hi,
> I new to this list but I hope that the list is what I'm after; an 
> implementation and design discussion list about SIP.
> If my question is wrongly addressed, please forgive me and please 
> point me the right direction...
>
> My question that I have searched all over for an answer to is quite
> simple:
>
> What is the correct behavior for a UA if a second INVITE arrives 
> before the first has been answered?
>
> SIP Flow:
>
> -------------> INVITE (1)
> <------------- 100 Trying (1)
> <------------- 180 Ringing (1)
> -------------> INVITE (2)
> ... ???
>
> Here the first (1) INVITE could have been answered by some other UA 
> that the INVITE might have been forked to and in this case session (2) 
> should start ringing.
>
>
> Niklas Fondberg
>
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Mon, 26 Mar 2007 23:01:13 +0100
> From: Adam Harding <adam.harding2 at ntlworld.com>
> Subject: [SIPForum-discussion] R-Factor type equation to evaluate VOIP 
> quality from Wireshark RTP stats
> To: "discussion at sipforum.org" <discussion at sipforum.org>
> Message-ID:
>
>
<20070326220113.LRUQ17393.aamtaout02-winn.ispmail.ntl.com at smtp.ntlworld.com>
>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hi,
>
> I am interested in any free algorithms that can be used to give a 
> value
for
> the voice quality in a VOIP call based on basic statistics such as 
> delay, packet loss and jitter.
>
> I asked this question a few days and got a really useful document 
> recommended to me which helps me understand how the R-Factor works but 
> I
can
> not get hold of the ITU-G values and my RTP results from Wireshark are 
> probably to basic to calculate the R-Factor.
>
> So just wondering if there is some sort of basic algorithm that I can
enter
> my results from the Wireshark RTP stats and get some sort of value of
voice
> quality that I can use to compare my results with each other.
>
> Thanks,
>
> Adam Harding
>
> -----------------------------------------
> Email sent from www.virginmedia.com/email Virus-checked using 
> McAfee(R) Software and scanned for spam
>
>
>
> ------------------------------
>
> Message: 3
> Date: Tue, 27 Mar 2007 09:50:17 +0800
> From: Deepanshu <deepanshu at huawei.com>
> Subject: Re: [SIPForum-discussion] simultaneous INVITEs
> To: Niklas Fondberg <niklas.fondberg at tilgin.com>
> Cc: discussion at sipforum.org
> Message-ID: <003801c77012$45a10050$8178a40a at china.huawei.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> inline
> ----- Original Message -----
> From: "Niklas Fondberg" <niklas.fondberg at tilgin.com>
> To: <discussion at sipforum.org>
> Sent: Tuesday, March 27, 2007 3:30 AM
> Subject: [SIPForum-discussion] simultaneous INVITEs
>
>
> > Hi,
> > I new to this list but I hope that the list is what I'm after; an 
> > implementation and design discussion list about SIP.
> > If my question is wrongly addressed, please forgive me and please 
> > point me the right direction...
> >
> > My question that I have searched all over for an answer to is quite
> > simple:
> >
> > What is the correct behavior for a UA if a second INVITE arrives 
> > before the first has been answered?
> >
> > SIP Flow:
> >
> > -------------> INVITE (1)
> > <------------- 100 Trying (1)
> > <------------- 180 Ringing (1)
> > -------------> INVITE (2)
> > ... ???
> >
> > Here the first (1) INVITE could have been answered by some other UA 
> > that the INVITE might have been forked to and in this case session 
> > (2) should start ringing.
>
> If the first (1) INVITE could have been answered by some other UA then 
> the originating UAC SHALL send a CANCEL request towards UAS instead of 
> a
INVITE
> (2).
>
> -------------> INVITE (1)
> <------------- 100 Trying (1)
> <------------- 180 Ringing (1)
> --------------> CANCEL (1)
>                         stop ringing
> <------------- SIP 487 (1)
>
>
> HTH
>
> Deepanshu Gautam
> R&D Engineer
> Huawei Technologies Co. Ltd.
> Nanjing, PRC
>
> >
> >
> > Niklas Fondberg
> >
> >
> >
> > _______________________________________________
> > This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or 
> > edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> > Post to the list at discussion at sipforum.org
> >
>
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 27 Mar 2007 08:05:23 -0700 (PDT)
> From: Sonja Belic <belic_sonja at yahoo.com>
> Subject: [SIPForum-discussion] Authentication and authorization in SIP
> To: discussion at sipforum.org
> Message-ID: <664094.34155.qm at web60623.mail.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>  I have a question regarding authentication and authorization 
> mechanism in SIP. For instance, if there are more then one 
> applications running on the same SIP system, does every application 
> authenticate itself or all of them use the same authentication parameters, defined for that SIP system?
>  Thanks in advance.
>
>  Best Regards,
>  Sonja
>
> ---------------------------------
> No need to miss a message. Get email on-the-go with Yahoo! Mail for 
> Mobile. Get started.
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> scrubbed...
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>
> ------------------------------
>
> _______________________________________________
> This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit 
> your delivery options, please visit
http://sipforum.
> org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
> End of discussion Digest, Vol 20, Issue 38
> ******************************************
>
>
> Powered by BYD Security Gateway.
>
>
>
> Powered by BYD Security Gateway.
>
>
>


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----


> _______________________________________________
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> your delivery options, please visit
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>




From sukerry at 126.com  Wed Mar 28 05:54:30 2007
From: sukerry at 126.com (sukerry)
Date: Wed, 28 Mar 2007 17:54:30 +0800
Subject: [SIPForum-discussion] Authentication and authorization in SIP
Message-ID: <460A3B9A.034C77.20049@m5-143.126.com>

Sonja Belic???????? 

????Every application authenticate itself

======== 2007-03-27 23:05:23 ???????????????? ========

Hi,
I have a question regarding authentication and authorization mechanism in SIP. For instance, if there are more then one applications running on the same SIP system, does every application authenticate itself or all of them use the same authentication parameters, defined for that SIP system?
Thanks in advance.

Best Regards,
Sonja


No need to miss a message. Get email on-the-go 
with Yahoo! Mail for Mobile. Get started. 

= = = = = = = = = = = = = = = = = = = = = = 
??????????????????
????

????????????????????????????sukerry
????????????????????????????sukerry at 126.com
??????????????????????????????2007-03-28
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From belic_sonja at yahoo.com  Wed Mar 28 06:01:02 2007
From: belic_sonja at yahoo.com (Sonja Belic)
Date: Wed, 28 Mar 2007 03:01:02 -0700 (PDT)
Subject: [SIPForum-discussion] Authentication and authorization in SIP
In-Reply-To: <664094.34155.qm@web60623.mail.yahoo.com>
Message-ID: <288514.96657.qm@web60616.mail.yahoo.com>

Hi, 
I'll try to explain my question in more details. If we have multihoming  ( more then one IP interface ) SIP system, with more then one client  applications, all implementing 
different services, and independently running on that SIP system, how 
should AA mechanism work? For instance, we use Digest auth scheme and 
receive challenge. The question is how shall we perform authentication, 
i.e. on which level? 
1. Should we have separate instance of Digest auth scheme for each 
application where each application controls own authentication parameters? 
2. Or we should consider just one Digest instance and set of 
authentication parameters performing authentication for system in whole 
(all applications shall use same, for example nonce, within their 
further SIP messages)? 
3. Or maybe we should take care about authentication per each interface 
of multihoming SIP system? 
 
This issue rise up the question about registration too. Shall we perform 
registration (SIP REGISTER request) for overall system at once or it 
should be done for each application separately? 
 
Are there any recommendations related to this issues? Is there any  dependency on the type of the network used? I would appreciate 
to get information about authentication practice in current SIP solutions. 
 
Thanks in advance. 
 
Best Regards, 
Sonja 

Sonja Belic <belic_sonja at yahoo.com> wrote: Hi,
 I have a question regarding authentication and authorization mechanism in SIP. For instance, if there are more then one applications running on the same SIP system, does every application authenticate itself or all of them use the same authentication parameters, defined for that SIP system?
 Thanks in advance.
 
 Best Regards,
 Sonja   

---------------------------------
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with Yahoo! Mail for Mobile. Get started._______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
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From rakesh_rcm at yahoo.com  Wed Mar 28 07:03:23 2007
From: rakesh_rcm at yahoo.com (rakesh menon)
Date: Wed, 28 Mar 2007 04:03:23 -0700 (PDT)
Subject: [SIPForum-discussion] One way speech
Message-ID: <256511.56654.qm@web56607.mail.re3.yahoo.com>

Hi all,

has anyone come accross "one way speech" issue.
mostly happens to external incomming calls.
PSTN to IP Phone.
Is it something to do with drops in RTP packets.

Regards,
Rakesh


 
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From gkittu at gmail.com  Wed Mar 28 08:34:54 2007
From: gkittu at gmail.com (Krishna Kishore G)
Date: Wed, 28 Mar 2007 18:04:54 +0530
Subject: [SIPForum-discussion] One way speech
In-Reply-To: <256511.56654.qm@web56607.mail.re3.yahoo.com>
References: <256511.56654.qm@web56607.mail.re3.yahoo.com>
Message-ID: <49f7a1c10703280534l6aada2a5w94e1d0334a452008@mail.gmail.com>

Hi  Rakesh,
       I am assuming this because reachability of IP(of assaigned to
IP phone) from mediagateway.From mediagw(assuming the debugging tool
prescence) try to ping the IP address of phone.
Aside  you can take ethereal traces at the gateway level, or at access
point to check out wether RTP stream is flowing out from mediagw

Regards
Kishore

On 3/28/07, rakesh menon <rakesh_rcm at yahoo.com> wrote:
> Hi all,
>
> has anyone come accross "one way speech" issue.
> mostly happens to external incomming calls.
> PSTN to IP Phone.
> Is it something to do with drops in RTP packets.
>
> Regards,
> Rakesh
>
>
>
> ____________________________________________________________________________________
> No need to miss a message. Get email on-the-go
> with Yahoo! Mail for Mobile. Get started.
> http://mobile.yahoo.com/mail
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>


-- 
Thanks &regards
G.Krishna Kishore

From rjsparks at nostrum.com  Wed Mar 28 10:23:37 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Wed, 28 Mar 2007 09:23:37 -0500
Subject: [SIPForum-discussion] Why Ack is different transaction?
In-Reply-To: <0fd201c77103$92b192a0$6c19320a@telxsi.com>
References: <63af059d0703272122l5134afpa51426b1551dd36@mail.gmail.com>
	<0fd201c77103$92b192a0$6c19320a@telxsi.com>
Message-ID: <B477B5F6-055D-47DF-A2C2-7CFEA9E3F65D@nostrum.com>

The destruction of the INVITE transaction on 200 is a known bug in  
the spec and a correction will be published shortly
(if you actually destroy the transaction, you will treat  
retransmissions a new request).

The real reason an ACK-200 is a new transaction is that it must  
follow the route-set established by the 200 to the INVITE.
In other words, it may go to a completely different first-hop  
destination than the INVITE did. Thus, it needs its own transaction
identifier.

RjS

On Mar 28, 2007, at 1:37 AM, Avishek Chowdhury wrote:

> Hi Aditi,
>
> When the UAC receives 200 OK, the client transaction is destroyed  
> at TL.
> This is done because, the TL is unaware of the above core. The  
> above core can be a Proxy or an UAC core.
> In case of proxy, the 200 OK is forwarded whereas in case of UAC,  
> an ACK is generated. Now this ACK has to create a new transaction  
> (transaction created by INVITE had been destroyed)
> at TL for its transmission, hence the ACK for 200 OK is outside the  
> INVITE transaction.
>
> For other non-2xx final responses, the client transaction at TL is  
> not destroyed and the ACK is generated by TL. Hence in this case,  
> the ACK is within the transaction.
>
> Regards,
> Avishek
>
> ----- Original Message -----
> From: aditi g
> To: discussion at sipforum.org
> Sent: Wednesday, March 28, 2007 9:52 AM
> Subject: [SIPForum-discussion] Why Ack is different transaction?
>
> Hi,
>
> I want to know why ACK is considered different transaction from  
> Invite transaction .
>
> regs
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit http:// 
> sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit http:// 
> sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org

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From rjsparks at nostrum.com  Wed Mar 28 10:48:40 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Wed, 28 Mar 2007 09:48:40 -0500
Subject: [SIPForum-discussion] SIPit 20 registration closes in 2 days
Message-ID: <160B4420-F4CB-441E-96D1-EEA960D20724@nostrum.com>

SIPit 20 registration closes this Friday, March 30 (2 days from today).

If you are planning to attend, but have not registered, please do so  
now.

The details for the event are available at www.sipit.net

You can register using this link: https://www.regonline.com/? 
eventID=123004&rTypeID=89030

See you in Antwerp!

RjS

From qt.kiran at gmail.com  Wed Mar 28 10:56:54 2007
From: qt.kiran at gmail.com (kiran chakkilam)
Date: Wed, 28 Mar 2007 20:26:54 +0530
Subject: [SIPForum-discussion] hi
Message-ID: <e0646ce70703280756j1155ac60i201d089e0e08e869@mail.gmail.com>

Hi all,

I have doubts on basic registration
                        UA                          Registrarserver
                      Register---------------->
                                 < --------------------200 Ok

1)In this Scenario whether Register request & 200 Ok called as a transaction
or not?

2)Whether it's possilble to send the Register request with out branch
parameter in VIA header?

3)are there any possiblities to send the register request with out VIA
header n Registration?

4)If i send aRequire  header contains INVITE , CANCEL in the Register
request to Registrar server(DUT) whether it responds with unsupported
parameter or not

5) If i am sending INVITE request toward Registrar (DUT) what are the
expected responses.



Thanks in Advance


Ch.kiran
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From nirk at MICROSOFT.com  Wed Mar 28 11:37:30 2007
From: nirk at MICROSOFT.com (Nir Katz)
Date: Wed, 28 Mar 2007 16:37:30 +0100
Subject: [SIPForum-discussion] SIP messages route
Message-ID: <59DD872D2D837D44B60E6B6C630CE4B2142403EF0F@EA-EXMSG-C303.europe.corp.microsoft.com>

How common is the scenario where the UAC and UAS sends SIP messages to one another directly after they have established the dialog and started media exchange? Or do most UAs continue to send the SIP messages using the same route they came from?   Does the protocol they use influence this behavior?

Thanks in advance

Nir Katz
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From indresh.singh at siemens.com  Wed Mar 28 14:05:39 2007
From: indresh.singh at siemens.com (Singh, Indresh (SNL US))
Date: Wed, 28 Mar 2007 11:05:39 -0700
Subject: [SIPForum-discussion] hi
In-Reply-To: <e0646ce70703280756j1155ac60i201d089e0e08e869@mail.gmail.com>
Message-ID: <3D80B10873C01D47BEC71C8DE311CF111CE07AC4@USNWK100MSX.ww017.siemens.net>

Below is my understanding. Hopefully it would help.
 
Regards,
 
Indresh

________________________________

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of kiran chakkilam
Sent: Wednesday, March 28, 2007 10:57 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] hi


Hi all,
 
I have doubts on basic registration
                        UA                          Registrarserver
                      Register---------------->
                                 < --------------------200 Ok
 
1)In this Scenario whether Register request & 200 Ok called as a
transaction or not?
[Singh, Indresh] Yes for devices compatible with RFC3261. 
 
2)Whether it's possilble to send the Register request with out branch
parameter in VIA header?
[Singh, Indresh] It is possible to send register request without branch
parameter in via header (if SIP device is compliant only with
RFC-2543/previous SIP RFC ). In that case the registrar server has to be
backward compatible with RFC-2543 to be able to process this request.
RFC-3261 recommends that SIP servers should be able to process requests
without branch parameter to maintain backward compatibility with
RFC-2543 
 
3)are there any possiblities to send the register request with out VIA
header n Registration?
[Singh, Indresh] No. Via header is mandatory in the requests. Refer to
Table 3 on page 163 of RFC-3261 Without via header in requests the
responses can not be sent. 
 
4)If i send aRequire  header contains INVITE , CANCEL in the Register
request to Registrar server(DUT) whether it responds with unsupported
parameter or not
[Singh, Indresh] Require Header or Allow header. Require header has tags
like timer ( indicating session timer support ) 100 rel ( indicating
PRACK support ) ?? 
 
5) If i am sending INVITE request toward Registrar (DUT) what are the
expected responses.
 Logically 405 Method Not allowed. Page 186
 
 
Thanks in Advance
 
 
Ch.kiran
 
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From parrishsteve2000 at yahoo.com  Wed Mar 28 14:58:05 2007
From: parrishsteve2000 at yahoo.com (Steve Parrish)
Date: Wed, 28 Mar 2007 13:58:05 -0500
Subject: [SIPForum-discussion] SIP messages route
In-Reply-To: <59DD872D2D837D44B60E6B6C630CE4B2142403EF0F@EA-EXMSG-C303.europe.corp.microsoft.com>
Message-ID: <008101c7716b$0660b930$031410ac@ibmhdqj6pzqq1l>

If you're talking about SIP endpoints that span subnets I would say it's
not very common at all due to NAT/firewall traversal.  Also with the
increasing number of SBC's (Session Border Controllers) being deployed I
would say that Service Providers force SIP traffic to hop along these
nodes.
 
Does the protocol they use influence this behavior?
Not really, there's no guarantee since the connection could be part UDP
and part TCP.
 
-Steve P.
 
 
-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Nir Katz
Sent: Wednesday, March 28, 2007 10:38 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] SIP messages route
 
How common is the scenario where the UAC and UAS sends SIP messages to
one another directly after they have established the dialog and started
media exchange? Or do most UAs continue to send the SIP messages using
the same route they came from?   Does the protocol they use influence
this behavior?
 
Thanks in advance
 
Nir Katz
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From mohandivakar2005 at yahoo.co.in  Thu Mar 29 03:43:27 2007
From: mohandivakar2005 at yahoo.co.in (mohan divakar)
Date: Thu, 29 Mar 2007 08:43:27 +0100 (BST)
Subject: [SIPForum-discussion] query regarding rtp loss in sip end points
Message-ID: <624344.55611.qm@web8606.mail.in.yahoo.com>

Hi,
  I have a question regarding rtp loss,
  there is a connection established between two sip users but both cant hear the voice of each other.
  I just want to know the reason why there is a loss of rtp when the connection is already established.
   
  thanks in advance
  Mohan
   

 				
---------------------------------
 Here?s a new way to find what you're looking for - Yahoo! Answers 
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From deepak.k at globaledgesoft.com  Thu Mar 29 07:09:47 2007
From: deepak.k at globaledgesoft.com (Deepak K)
Date: Thu, 29 Mar 2007 16:39:47 +0530
Subject: [SIPForum-discussion] SIPit 20 : what's about IMS Interop?
References: <3D80B10873C01D47BEC71C8DE311CF111CE07AC4@USNWK100MSX.ww017.siemens.net>
Message-ID: <00c701c771f2$c3727130$520710ac@globaledgesoft.com>

HI,

SIP is a major element of the IMS architecture.

Wanted to know whether IMS/3GPP specific SIP extension/implementations interop would be one of the major focus point of this event ? 

Regards,
DK
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From sipcbi at yahoo.co.in  Thu Mar 29 08:42:09 2007
From: sipcbi at yahoo.co.in (sip cbi)
Date: Thu, 29 Mar 2007 13:42:09 +0100 (BST)
Subject: [SIPForum-discussion] can use jsr 281  for creating SIP phone??
Message-ID: <957311.22569.qm@web94312.mail.in2.yahoo.com>

Dear All,
   
  Greetings!!!
   
  can we create SIP Phone (User Agent) using JSR 281 through Java 2 standard Edition ?
   
  Thanks,
  sipcbi

 				
---------------------------------
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From oguzhan at nevotek.com  Thu Mar 29 09:12:16 2007
From: oguzhan at nevotek.com (Oguzhan Cem)
Date: Thu, 29 Mar 2007 16:12:16 +0300
Subject: [SIPForum-discussion] can use jsr 281 for creating SIP phone??
Message-ID: <6F0695B2993AAF4EB5A63AEA95A44BBFF9386A@buddy.nevotek.com>

Hello,

 

I also want to ask a question related,

 

If we create a sip agent, (with any platform including JSR 281) does
anyone give me any idea how to test it? Any test site installed for this
purpose?

 

Thx in advance,

 

Oguzhan Cem.

 

 

 

  _____  

From: sip cbi [mailto:sipcbi at yahoo.co.in] 
Sent: Thursday, March 29, 2007 3:42 PM
To: 
Subject: [SIPForum-discussion] can use jsr 281 for creating SIP phone??

 

Dear All,

 

Greetings!!!

 

can we create SIP Phone (User Agent) using JSR 281 through Java 2
standard Edition ?

 

Thanks,

sipcbi

  

  _____  

Here's a new way to find what you're looking for - Yahoo! Answers
<http://us.rd.yahoo.com/mail/in/yanswers/*http:/in.answers.yahoo.com/>  

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From indresh.singh at siemens.com  Thu Mar 29 16:41:49 2007
From: indresh.singh at siemens.com (Singh, Indresh (SNL US))
Date: Thu, 29 Mar 2007 13:41:49 -0700
Subject: [SIPForum-discussion] query regarding rtp loss in sip end points
In-Reply-To: <624344.55611.qm@web8606.mail.in.yahoo.com>
Message-ID: <3D80B10873C01D47BEC71C8DE311CF111D1CA179@USNWK100MSX.ww017.siemens.net>

SIP does only the signaling and during the signaling the SDP/media is
exchanged between the two devices using SIP Signaling protocol. If the
SDP exchanged between two devices do not properly exchange
 
1) codecs
2) RTP port and addresses where they will send and receive media
 
You will not have a speech path. This could be one reason why you do not
have speec path. There could be other reasons as well....
 
Regards,
 
Indresh K Singh

________________________________

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of mohan divakar
Sent: Thursday, March 29, 2007 3:43 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] query regarding rtp loss in sip end
points


Hi,
I have a question regarding rtp loss,
there is a connection established between two sip users but both cant
hear the voice of each other.
I just want to know the reason why there is a loss of rtp when the
connection is already established.
 
thanks in advance
Mohan
 

________________________________

Here's a new way to find what you're looking for - Yahoo! Answers
<http://us.rd.yahoo.com/mail/in/yanswers/*http://in.answers.yahoo.com/>

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From nvvgopal80 at rediffmail.com  Thu Mar 29 22:43:19 2007
From: nvvgopal80 at rediffmail.com (venkata venu gopal)
Date: 30 Mar 2007 02:43:19 -0000
Subject: [SIPForum-discussion] can use jsr 281 for creating SIP phone??
Message-ID: <20070330024319.1675.qmail@webmail99.rediffmail.com>

Hello,

There are test frameworks for the SIP to test the device for the confirmance of the RFCs, some torture test cases and interoperability and etc. Find below the URLs to find the test framework and the test suites.. 

Test Framework : http://www.sipfoundry.org/sip-forum-test-framework/sip-forum-test-framework-sftf.html

Test Suites :  http://www.iol.unh.edu/services/testing/voip/testsuites/#SIP%20Conformance%20Test%20Suite

There may be other sites better then this but any information in this direction are welcomed and appreciated.

Hope this is helpful to you.. 

Thanks,
Venu. ?


On Thu, 29 Mar 2007 Oguzhan Cem wrote :
>Hello,
>
>
>
>I also want to ask a question related,
>
>
>
>If we create a sip agent, (with any platform including JSR 281) does
>anyone give me any idea how to test it? Any test site installed for this
>purpose?
>
>
>
>Thx in advance,
>
>
>
>Oguzhan Cem.
>
>
>
>
>
>
>
>   _____
>
> From: sip cbi [mailto:sipcbi at yahoo.co.in]
>Sent: Thursday, March 29, 2007 3:42 PM
>To:
>Subject: [SIPForum-discussion] can use jsr 281 for creating SIP phone??
>
>
>
>Dear All,
>
>
>
>Greetings!!!
>
>
>
>can we create SIP Phone (User Agent) using JSR 281 through Java 2
>standard Edition ?
>
>
>
>Thanks,
>
>sipcbi
>
>
>
>   _____
>
>Here's a new way to find what you're looking for - Yahoo! Answers
><http://us.rd.yahoo.com/mail/in/yanswers/*http:/in.answers.yahoo.com/>
>
>_______________________________________________
>This is the SIP Forum discussion mailing list
>TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
>Post to the list at discussion at sipforum.org
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From rjsparks at nostrum.com  Fri Mar 30 00:21:34 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Thu, 29 Mar 2007 23:21:34 -0500
Subject: [SIPForum-discussion] SIPit 20 registration closes today (March 30)
Message-ID: <E2CAE4B2-6A2C-434B-BD09-5E5799300B04@nostrum.com>

Registration for SIPit 20 closes today (March 30).

If you plan to attend, but have not yet registered, do so immediately.

Information on the event and the link for registration can be found  
at www.sipit.net.

See you in Antwerp!

RjS

From invite at friends.unicefusa.org  Fri Mar 30 06:09:45 2007
From: invite at friends.unicefusa.org (gujjenaveen@gmail.com)
Date: Fri, 30 Mar 2007 03:09:45 -0700
Subject: [SIPForum-discussion] Inviting my friends & family...
Message-ID: <1175249385.27076@unicef.popularmediamail.org>

I'm extending a personal invitation to my friends and family to make a difference without spending a penny.  To see your invitation, click the link below, or copy and paste it into your browser's address field:

http://friends.unicefusa.org/r/6eaf7b3c2fdf102a8325











If you would prefer not to receive invitations from Friends.UNICEFUSA.org please click here 
http://friends.unicefusa.org/?PC=UNSUB&rh=22ce585c3ac9ab673813db0da6408e93&sender=naveeng at intoto.com&tc=12

----------------------------------------------------------
UNICEF USA
PMB# 210
2440 16th Street
San Francisco, CA 94103-4211


From hariprasad.taduru at gmail.com  Fri Mar 30 05:20:20 2007
From: hariprasad.taduru at gmail.com (Taduru Hariprasad)
Date: Fri, 30 Mar 2007 14:50:20 +0530
Subject: [SIPForum-discussion] Hi
Message-ID: <4dff78790703300220v4b08a07cm5b17f502ecbade89@mail.gmail.com>

Hi,

I started to learn sip. Currently going thru rfc 3261. Can i have th
answers for the
following doubts.

1) BRANCH and CSeq parameters will get change for every transaction?

2) How Ack is treated as one transaction if the final response is 200-ok for
   an INVITE?

3) How to detect loops and overcome them?

And also please mension/attach the docs if you have for my referrence.

Thanks
Hari

From sipcbi at yahoo.co.in  Fri Mar 30 06:45:55 2007
From: sipcbi at yahoo.co.in (sip cbi)
Date: Fri, 30 Mar 2007 11:45:55 +0100 (BST)
Subject: [SIPForum-discussion] how to create java installable ?????
Message-ID: <278653.85848.qm@web94304.mail.in2.yahoo.com>

Hello All !!!!

how to create java exe file and installable????
is there any tool to make java exe ....
please give me the site to download the tool....

Regards
sipcbi

 				
---------------------------------
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From indresh.singh at siemens.com  Fri Mar 30 10:58:11 2007
From: indresh.singh at siemens.com (Singh, Indresh (SNL US))
Date: Fri, 30 Mar 2007 07:58:11 -0700
Subject: [SIPForum-discussion] Hi
In-Reply-To: <4dff78790703300220v4b08a07cm5b17f502ecbade89@mail.gmail.com>
Message-ID: <3D80B10873C01D47BEC71C8DE311CF111D1CA47A@USNWK100MSX.ww017.siemens.net>

 

-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Taduru Hariprasad
Sent: Friday, March 30, 2007 5:20 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] Hi

Hi,

I started to learn sip. Currently going thru rfc 3261. Can i have th
answers for the
following doubts.

1) BRANCH and CSeq parameters will get change for every transaction?

>> Yes. 

2) How Ack is treated as one transaction if the final response is 200-ok
for
   an INVITE?

>> Yes. But it is a unique transaction and it's cseq is same as
INVITE-200OK. Just the branch-identifier is different.

3) How to detect loops and overcome them?

>> By checking if the ipAddress of your box is there in one of the i/c
via headers. But after this you have to differentiate between loop and
spiraling before you can say loop detected. 

And also please mension/attach the docs if you have for my referrence.

>> Everything in RFC-3261 :)

Thanks
Hari
_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org


From hsingh at wgate.com  Fri Mar 30 11:41:37 2007
From: hsingh at wgate.com (Harbinder Singh)
Date: Fri, 30 Mar 2007 11:41:37 -0400
Subject: [SIPForum-discussion] Hi
In-Reply-To: <3D80B10873C01D47BEC71C8DE311CF111D1CA47A@USNWK100MSX.ww017.siemens.net>
Message-ID: <FA7EECCA3735EF46BE49245D249B788CED6B25@exchange1.wgate.com>

Also, more explanation in line:--

-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Singh, Indresh
(SNL US)
Sent: Friday, March 30, 2007 10:58 AM
To: Taduru Hariprasad; discussion at sipforum.org
Subject: Re: [SIPForum-discussion] Hi

 

-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Taduru Hariprasad
Sent: Friday, March 30, 2007 5:20 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] Hi

Hi,

I started to learn sip. Currently going thru rfc 3261. Can i have th
answers for the
following doubts.

1) BRANCH and CSeq parameters will get change for every transaction?

>> Yes. 
----For example, in a call: Each successive request during a call will
have a higher CSeq number. Also, the caller and the called parties each
maintain their own separate Cseq counts.

2) How Ack is treated as one transaction if the final response is 200-ok
for
   an INVITE?

>> Yes. But it is a unique transaction and it's cseq is same as
INVITE-200OK. Just the branch-identifier is different.

---- INVITE is the only method in SIP in which there is this three-way
handshake involving ACK. All other SIP requests are of the form
REQUEST/200 or REQUEST/4xx or 5xx or 6xx for a failure. That is why ACK
has the same Cseq number as the other two - INVITE and 200.

3) How to detect loops and overcome them?

>> By checking if the ipAddress of your box is there in one of the i/c
via headers. But after this you have to differentiate between loop and
spiraling before you can say loop detected. 

And also please mension/attach the docs if you have for my referrence.

>> Everything in RFC-3261 :)

Thanks
Hari
_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org

_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org



From suryaprakashu at rediffmail.com  Fri Mar 30 12:07:49 2007
From: suryaprakashu at rediffmail.com (Surya  Prakash Ummadi)
Date: 30 Mar 2007 16:07:49 -0000
Subject: [SIPForum-discussion] Hi
Message-ID: <20070330160749.26122.qmail@webmail36.rediffmail.com>

 ?
Hi,

   please comment below  following  answers :


1) BRANCH and CSeq parameters will get change for every transaction?
>
> >> Yes.(this is correct)
>
>2) How Ack is treated as one transaction if the final response is 200-ok
>for
>    an INVITE?
>
> >> Yes. But it is a unique transaction and it's cseq is same as
>INVITE-200OK. Just the branch-identifier is different.
>
>3) How to detect loops and overcome them?
>
> >> For detecting loops ,branch-id is used as the reference.Branch-id is formed by the magiccooke and hash of the request-uri,from tag,totag ,cseq,callid and topmost VIA header.IF it is not the loop,atleast request-uri will change.So branch-id will be different.IF it same,then loop is detected.

Another way is MAX-Forwards.
>
>And also please mension/attach the docs if you have for my referrence.

  refer to the tech-invite site,where u can find good presentation of documents.
>
On Fri, 30 Mar 2007 Singh,Indresh(SNL US) wrote :
>
>

regards
surya
>-----Original Message-----
> From: discussion-bounces at sipforum.org
>[mailto:discussion-bounces at sipforum.org] On Behalf Of Taduru Hariprasad
>Sent: Friday, March 30, 2007 5:20 AM
>To: discussion at sipforum.org
>Subject: [SIPForum-discussion] Hi
>
>Hi,
>
>I started to learn sip. Currently going thru rfc 3261. Can i have th
>answers for the
>following doubts.
>
>1) BRANCH and CSeq parameters will get change for every transaction?
>
> >> Yes.
>
>2) How Ack is treated as one transaction if the final response is 200-ok
>for
>    an INVITE?
>
> >> Yes. But it is a unique transaction and it's cseq is same as
>INVITE-200OK. Just the branch-identifier is different.
>
>3) How to detect loops and overcome them?
>
> >> By checking if the ipAddress of your box is there in one of the i/c
>via headers. But after this you have to differentiate between loop and
>spiraling before you can say loop detected.
>
>And also please mension/attach the docs if you have for my referrence.
>
> >> Everything in RFC-3261 :)
>
>Thanks
>Hari
>_______________________________________________
>This is the SIP Forum discussion mailing list
>TO UNSUBSCRIBE, or edit your delivery options, please visit
>http://sipforum.org/mailman/listinfo/discussion
>Post to the list at discussion at sipforum.org
>
>_______________________________________________
>This is the SIP Forum discussion mailing list
>TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
>Post to the list at discussion at sipforum.org
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From tkishor at softlinkindia.com  Fri Mar 30 23:07:42 2007
From: tkishor at softlinkindia.com (Kishor)
Date: Sat, 31 Mar 2007 08:37:42 +0530
Subject: [SIPForum-discussion] (no subject)
Message-ID: <001601c77341$ea265e40$d728e0dc@qpo2pp>


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From am2866 at columbia.edu  Sat Mar 31 04:27:03 2007
From: am2866 at columbia.edu (Arpit Mehta)
Date: Sat, 31 Mar 2007 04:27:03 -0400
Subject: [SIPForum-discussion] Regarding Caller ID
Message-ID: <bbd5fc140703310127q4f5e643fhaae3a2566cf597bd@mail.gmail.com>

Hello,

I have a problem regarding Caller ID. I am running the SipC client on my
machine. I am using a Cisco 2600 router. I have configured the gateway so as
to connect to my SipC client when a call is made to the gateway. Now when I
make a call, it connects to the SipC and everything works fine. But I need
the number of the person who has called SipC. In SipC also it displays
unknown and does not display the number. Are there any possible reasons for
the number not being shown? Does it look to be a configuration problem at
the router side so that it is not getting the caller's ID?

Any suggestion would be helpful.

Thanks.

--
Arpit
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From umair3210 at yahoo.com  Sat Mar 31 08:07:33 2007
From: umair3210 at yahoo.com (Muhammad Umair)
Date: Sat, 31 Mar 2007 05:07:33 -0700 (PDT)
Subject: [SIPForum-discussion] programming SIP??? help me
Message-ID: <117401.15464.qm@web38711.mail.mud.yahoo.com>

hi all,
   
  i have done the reading about the SIP. can any one tell me how can programatically (using any languange C#.net,vb.net) implement SIP.
   
  thanx in advance
  Regards
  Muhammad Umair

 
---------------------------------
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with Yahoo! Mail for Mobile. Get started.
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From jani_tech_forum at yahoo.com  Sat Mar 31 10:13:47 2007
From: jani_tech_forum at yahoo.com (Janakiraman N)
Date: Sat, 31 Mar 2007 07:13:47 -0700 (PDT)
Subject: [SIPForum-discussion] programming SIP??? help me
In-Reply-To: <117401.15464.qm@web38711.mail.mud.yahoo.com>
Message-ID: <562500.36371.qm@web62111.mail.re1.yahoo.com>

Hi Muhammad Umair,

You can implement your SIP knowledge using SIP Servlet which will be used to create a SIP service. Its basically JAVA.

If you want to know in detail, please read JSR 116 standard SIP Servlet

Regards,
Janakiraman. N

Muhammad Umair <umair3210 at yahoo.com> wrote: hi all,
   
  i have done the reading about the SIP. can any one tell me how can programatically (using any languange C#.net,vb.net) implement SIP.
   
  thanx in advance
  Regards
  Muhammad Umair
   

---------------------------------
No need to miss a message. Get email on-the-go 
with Yahoo! Mail for Mobile. Get started._______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org


 
---------------------------------
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Get your game face on with the latest PS3 news and previews at Yahoo! Games.
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From karim_balkas at hotmail.com  Sat Mar 31 14:49:10 2007
From: karim_balkas at hotmail.com (balkas karim)
Date: Sat, 31 Mar 2007 18:49:10 +0000
Subject: [SIPForum-discussion] simulate sip with NS2
Message-ID: <BAY12-F22A1E34D54AB6DA3624EF8FA620@phx.gbl>

hi all,

if you can help me about simulation the protocol SIP with Network Simulator 
2 (NS2)!!!

thanks for response!!

karim

_________________________________________________________________
MSN Messenger: appels gratuits de PC ? PC ! 
http://www.msn.fr/newhotmail/Default.asp?Ath=f


From g.shashira at gmail.com  Sat Mar 31 23:54:50 2007
From: g.shashira at gmail.com (shashira guntuka)
Date: Sun, 1 Apr 2007 09:24:50 +0530
Subject: [SIPForum-discussion] SIP training in Hyderabad
Message-ID: <35ed02cf0703312054s68dad9e5te624c894d355721e@mail.gmail.com>

Hi All,

New to this group and would like to find out if there are any training
centres in Hyderabad that teach VoIP and SIP.

regards,
Shashira.
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From deepaknivas at rediffmail.com  Thu Mar  1 03:10:26 2007
From: deepaknivas at rediffmail.com (Deepak nivas)
Date: 1 Mar 2007 03:10:26 -0000
Subject: [SIPForum-discussion] query
Message-ID: <20070301031026.11389.qmail@webmail90.rediffmail.com>

  
Hi,



can any one explain potential race condition in sip?



Thanks in advance.

Regards,
Deepak.
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From deepaknivas at rediffmail.com  Thu Mar  1 10:06:57 2007
From: deepaknivas at rediffmail.com (Deepak nivas)
Date: 1 Mar 2007 10:06:57 -0000
Subject: [SIPForum-discussion] query
Message-ID: <20070301100657.2816.qmail@webmail99.rediffmail.com>

  Hi,


 What is mean by loose routing?


Regards,
Deepak.
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From williamp at TechMahindra.com  Thu Mar  1 11:18:23 2007
From: williamp at TechMahindra.com (William Prusty)
Date: Thu, 1 Mar 2007 16:48:23 +0530
Subject: [SIPForum-discussion] loose routing
Message-ID: <5C5A863D4858FC41818EF1C777DB6CFA033CD030@SINBNGEX001.TechMahindra.com>


 

 

Strict Routing and Loose Routing

The way how record routing works has evolved. Record routing according
to RFC2543 rewrote the Request-URIi
<http://www.iptel.org/glossary#term81> i
<http://www.iptel.org/glossary#term80> . That means the Request-URI
always
contained URI of the next hop (which can be either next proxy server
which inserted Record-Route header field or destination user agent).
Because of that it was necessary to save the original Request-URI as the
last Route header field. This approach is called strict routing.

Loose routing, as specified in RFC3261, works in a little bit different
way. The Request-URI is no more overwritten, it always contains URI of
the destination user agent. If there are any Route header field in a
message, than the message is sent to the URI from the topmost Route
header field. This is significant change--Request-URI doesn't
necessarily contain URI to which the request will be sent. In fact,
loose routing is very similar to IP source routing.

Because transit from strict routing to loose routing would break
backwards compatibility and older user agents wouldn't work, it is
necessary to make loose routing backwards compatible. The backwards
compatibility unfortunately adds a lot of overhead and is often source
of major problems.

Regards,

william

 



============================================================================================================================

Tech Mahindra, formerly Mahindra-British Telecom.
 
Disclaimer:

This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review at <a href="http://www.techmahindra.com/Disclaimer.html">http://www.techmahindra.com/Disclaimer.html</a> externally and <a href="http://tim.techmahindra.com/Disclaimer.html">http://tim.techmahindra.com/Disclaimer.html</a> internally within Tech Mahindra.

============================================================================================================================
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From NXDR43 at motorola.com  Thu Mar  1 11:51:28 2007
From: NXDR43 at motorola.com (Dawn Somen-NXDR43)
Date: Thu, 1 Mar 2007 19:51:28 +0800
Subject: [SIPForum-discussion] query
Message-ID: <40E89886C8B3B54B98C5291646C591AA014B6206@ZMY16EXM67.ds.mot.com>

HI!
 
Can someone tell me if no expiration is present, is it 3600sec by
default or the server chooses a time interval for that as stated in sec
10.2.1.1 of RFC3261?
Also, is 360 configured locally?
 
Thanks!
Regards,
Somen
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From nirk at MICROSOFT.com  Thu Mar  1 12:58:28 2007
From: nirk at MICROSOFT.com (Nir Katz)
Date: Thu, 1 Mar 2007 12:58:28 +0000
Subject: [SIPForum-discussion] Interoperability Testing - Best course of
	action
Message-ID: <59DD872D2D837D44B60E6B6C630CE4B212D7A8CB68@EA-EXMSG-C303.europe.corp.microsoft.com>

Hi,

Based on your experience what is the best way to test SIP Application Layer Gateway ability to work with other (especially hardware) SIP solutions? Is there any sense in trying to automate the work with different hardware? Or should the focus be on RFC compliance with sporadic and direct manual testing of the various SIP solutions interoperability?

Thanks in advance

Nir Katz



From francesco.landolfo at gmail.com  Thu Mar  1 14:14:23 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Thu, 1 Mar 2007 15:14:23 +0100
Subject: [SIPForum-discussion] About Header Contact
In-Reply-To: <d08f948c0703010608x462c0f39u4f4ba351e143fc37@mail.gmail.com>
References: <d08f948c0703010608x462c0f39u4f4ba351e143fc37@mail.gmail.com>
Message-ID: <d08f948c0703010614g5c962a46s4de2a21fb0bbb1d1@mail.gmail.com>

Hi,
I have the following doubt.
Suppose that you have two client:

   - A (sip:a at mydomain.org);
   - B ( sip:b at mydomain.org).

Suppose you have a Sip Proxy Server too.

I want to setup a call session using Sip Proxy Server. (Please see the
attached file)

When B sends 180 RINGING back to the client A, does it writes in the Contact
header its sip address  (Contact: "B" < sip:B at mydomain.org>) or its address
of care (Contact: "A" < sip:B at 100.101.102.103>) ???

What is the standard method?

Thanks,
Francesco Paolo Landolfo

-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
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From francesco.landolfo at gmail.com  Thu Mar  1 14:08:37 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Thu, 1 Mar 2007 15:08:37 +0100
Subject: [SIPForum-discussion] About Header Contact
Message-ID: <d08f948c0703010608x462c0f39u4f4ba351e143fc37@mail.gmail.com>

Hi,
I have the following doubt.
Suppose that you have two client:

   - A (sip:a at mydomain.org);
   - B ( sip:b at mydomain.org).

Suppose you have a Sip Proxy Server too.

I want to setup a call session using Sip Proxy Server. (Please see the
attached file)

When B sends 180 RINGING back to the client A, does it writes in the Contact
header its sip address  (Contact: "B" < sip:B at mydomain.org>) or its address
of care (Contact: "A" <sip:B at 100.101.102.103>) ???

What is the standard method?

Thanks,
Francesco Paolo Landolfo

-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
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From deepaknivas at rediffmail.com  Fri Mar  2 03:14:30 2007
From: deepaknivas at rediffmail.com (Deepak nivas)
Date: 2 Mar 2007 03:14:30 -0000
Subject: [SIPForum-discussion] query
Message-ID: <20070302031430.13857.qmail@webmail106.rediffmail.com>

  
Hi,


Can any explain conference call flow with a diagram?


Regards,
Deepak.

Regards,
Deepak.
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From zeroroot at tmax.co.kr  Fri Mar  2 08:49:52 2007
From: zeroroot at tmax.co.kr (Young-Geun Park)
Date: Fri, 2 Mar 2007 17:49:52 +0900
Subject: [SIPForum-discussion]  recommanding a tester for sip server
Message-ID: <000001c75ca7$bebeaf10$3d01a8c0@zeroroot>

How about SIPp(sipp.sourceforge.net) for a sip tester?

 

Regards,

Park

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From nicolas.destor at orange-ftgroup.com  Fri Mar  2 14:51:36 2007
From: nicolas.destor at orange-ftgroup.com (zze-DESTOR Nicolas RD-SIRP-LAN)
Date: Fri, 2 Mar 2007 15:51:36 +0100
Subject: [SIPForum-discussion] query
In-Reply-To: <20070302031430.13857.qmail@webmail106.rediffmail.com>
Message-ID: <BAF83494CE653943A97B9F755016A06604A8661C@ftrdmel1.rd.francetelecom.fr>

Hi, 
 
Does someone know a very simple open-source voice-mail server in JAVA or
C++ ( GUI and record functionnality is not nescessary)? 
It's to do somes modifications on it after, but I'm not an expert
programmer so I'm looking for a simple voice-mail before begin coding! 
 
thanks for your help. (I don't ask to you Kaushik, you help me already a
lot!)
 
regards,
Nicolas 
 
 
For information, here the call-flow that the modified voice-mail need to
support. 
 
 
 
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From durgani at gmail.com  Fri Mar  2 15:13:02 2007
From: durgani at gmail.com (Prakash Durgani)
Date: Fri, 2 Mar 2007 10:13:02 -0500
Subject: [SIPForum-discussion] query
In-Reply-To: <BAF83494CE653943A97B9F755016A06604A8661C@ftrdmel1.rd.francetelecom.fr>
References: <20070302031430.13857.qmail@webmail106.rediffmail.com>
	<BAF83494CE653943A97B9F755016A06604A8661C@ftrdmel1.rd.francetelecom.fr>
Message-ID: <f6f514070703020713t5a2b30eawcfa2aeca882a639f@mail.gmail.com>

Have you looked at Asterisk? or maybe a combination of SER (SIP Express
Router) and SEMS (SIP Express Media Server).

On 3/2/07, zze-DESTOR Nicolas RD-SIRP-LAN <nicolas.destor at orange-ftgroup.com>
wrote:
>
>  Hi,
>
> Does someone know a very simple open-source voice-mail server in JAVA or
> C++ ( GUI and record functionnality is not nescessary)?
> It's to do somes modifications on it after, but I'm not an
> expert programmer so I'm looking for a simple
> voice-mail before begin coding!
>
> thanks for your help. (I don't ask to you Kaushik, you help me already a
> lot!)
>
> regards,
> Nicolas
>
>
> For information, here the call-flow that the modified voice-mail need to
> support.
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
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From sivam at motorola.com  Fri Mar  2 15:39:51 2007
From: sivam at motorola.com (Siva M-Q16748)
Date: Fri, 2 Mar 2007 23:39:51 +0800
Subject: [SIPForum-discussion] query
In-Reply-To: <20070301100657.2816.qmail@webmail99.rediffmail.com>
Message-ID: <988EE2C769AC284ABAE9328BFC10703F01815135@ZMY16EXM66.ds.mot.com>

Hi
 
In case of loose routing in the responce sent to a request ,Request-URI
would contain the final destination to be reached and the Route header
is used as the path to reach the same
 
Where as in strict routing the Request-URI would say the next hop to be
reached and the last entry in route will be the final destination to be
reached
 
There are very good examples in RFC3261 Section 16.12
 
Siva

________________________________

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Deepak nivas
Sent: Thursday, March 01, 2007 3:37 PM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] query



  Hi,


What is mean by loose routing?


Regards,
Deepak. 



 
<http://adworks.rediff.com/cgi-bin/AdWorks/sigclick.cgi/www.rediff.com/s
ignature-home.htm/1507191490 at Middle5?PARTNER=3>  
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From nicolas.destor at orange-ftgroup.com  Fri Mar  2 16:12:26 2007
From: nicolas.destor at orange-ftgroup.com (zze-DESTOR Nicolas RD-SIRP-LAN)
Date: Fri, 2 Mar 2007 17:12:26 +0100
Subject: [SIPForum-discussion] query
In-Reply-To: <f6f514070703020713t5a2b30eawcfa2aeca882a639f@mail.gmail.com>
Message-ID: <BAF83494CE653943A97B9F755016A06604A86699@ftrdmel1.rd.francetelecom.fr>

Sorry for my late answer.
Yes I looked at Asterisk but It's not what I want. The voice-mail will be install on a existing SIP network (the sip server is already present). In fact the installation process have to be the same than a sipphone, the alone difference is that the voice-mail answer automaticelly when it receive a INVITE message!  
I don't know SER and SEMS but I think is the same problem...
 
But thanks for your response! If you have anothers ideas says me !
 


________________________________

De : Prakash Durgani [mailto:durgani at gmail.com] 
Envoy? : vendredi 2 mars 2007 16:13
? : zze-DESTOR Nicolas RD-SIRP-LAN
Cc : discussion at sipforum.org
Objet : Re: [SIPForum-discussion] query


Have you looked at Asterisk? or maybe a combination of SER (SIP Express Router) and SEMS (SIP Express Media Server).


On 3/2/07, zze-DESTOR Nicolas RD-SIRP-LAN <nicolas.destor at orange-ftgroup.com> wrote: 

	Hi, 
	 
	Does someone know a very simple open-source voice-mail server in JAVA or C++ ( GUI and record functionnality is not nescessary)? 
	It's to do somes modifications on it after, but I'm not an expert programmer so I'm looking for a simple voice-mail before begin coding! 
	 
	thanks for your help. (I don't ask to you Kaushik, you help me already a lot!)
	 
	regards,
	Nicolas 
	 
	 
	For information, here the call-flow that the modified voice-mail need to support. 
	 
	
	_______________________________________________
	This is the SIP Forum discussion mailing list
	TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
	Post to the list at discussion at sipforum.org
	
	


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From sicu13 at yahoo.ca  Fri Mar  2 23:16:21 2007
From: sicu13 at yahoo.ca (Sicu Babanul)
Date: Fri, 2 Mar 2007 18:16:21 -0500 (EST)
Subject: [SIPForum-discussion] virtual number forward to my cell
Message-ID: <919882.46090.qm@web63103.mail.re1.yahoo.com>

  I am trying to find a virtual number from Romania that I can forward to my mobile, does anyone know where I can get one?
thank you

  
 		
---------------------------------
Be smarter than spam. See how smart SpamGuard is at giving junk email the boot with the All-new Yahoo! Mail  
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From naveed770 at yahoo.com  Sat Mar  3 06:34:48 2007
From: naveed770 at yahoo.com (naveed khan)
Date: Fri, 2 Mar 2007 22:34:48 -0800 (PST)
Subject: [SIPForum-discussion] Authentication for third party registration
	in SIP
Message-ID: <638333.897.qm@web60019.mail.yahoo.com>

hi to all
   
  Can any one of you tell me that how the authentication is done in case of third party registration.
  And I need your help regarding why there is a need for third party registration in sip.
  How a third party (supposed Bob) will come to know that he has to register on the behalf of say "Alice".
  Is there any method suggessted by ietf for the authentication of third party registration.
  And who is to authenticate in this process either third person in my case "Bob" or the "Alice".
   
  Thanks in advance for any kind of help
   
  Regards,
  Naveed Khan
   

 
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From ext.lore.matheau at sncf.fr  Mon Mar  5 08:20:02 2007
From: ext.lore.matheau at sncf.fr (EXT / LORE MATHEAU Franck)
Date: Mon, 5 Mar 2007 09:20:02 +0100
Subject: [SIPForum-discussion] RE :  query
In-Reply-To: <20070302031430.13857.qmail@webmail106.rediffmail.com>
Message-ID: <00DC188807DB90449DD9384493C07F787AC150@s72sdeig073.ig.sncf.fr>

Hy all,

For any call flow that yuo need, follow this link :

 

http://www.tech-invite.com/

 

Franck

 

-----Message d'origine-----
De : discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] 
Envoy? : vendredi 2 mars 2007 04:15
? : discussion
Objet : [SIPForum-discussion] query

 

  
Hi,


Can any explain conference call flow with a diagram?


Regards,
Deepak. 

Regards, 
Deepak.

 <http://adworks.rediff.com/cgi-bin/AdWorks/sigclick.cgi/www.rediff.com/signature-home.htm/1507191490 at Middle5?PARTNER=3> 

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From deepaknivas at rediffmail.com  Mon Mar  5 09:25:59 2007
From: deepaknivas at rediffmail.com (Deepak nivas)
Date: 5 Mar 2007 09:25:59 -0000
Subject: [SIPForum-discussion] query
Message-ID: <20070305092559.5005.qmail@webmail90.rediffmail.com>

  hi,

what is use In-Reply-To header field?


regards,
deepak.



Regards,
Deepak.
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From imyousuf at gmail.com  Tue Mar  6 04:38:30 2007
From: imyousuf at gmail.com (Imran M Yousuf)
Date: Tue, 6 Mar 2007 10:38:30 +0600
Subject: [SIPForum-discussion] About SIP Presence
Message-ID: <7bfdc29a0703052038i7f3ab2dp4a446592d3a5a733@mail.gmail.com>

Dear Forum members,

I have a query regarding SIP Presence flow. I will illustrate my question
with an example.

imran at smartitengineering.com REQUEST to SUBSCRIBE to
imyousuf at smartitengineering.com EVENT: presence
PROXY - Synchronize with NOTIFY to - imran at smartitengineering.com
Now my question is how does imyousuf at smartitengineering.com update his
information to the PROXY?
As far as my understanding goes "imyousuf" will PUBLISH his status to the
PROXY; now if the UA of "imyousuf" does not ALLOW PUBLISH, can the PROXY
REQUEST to SUBSCRIBE to imyousuf at smartitengineering.com EVENT: presence?

Thanks in advance,

Imran M Yousuf
Enterpreneur & Lead Developer
Smart IT Engineering
Dhaka, Bangladesh
Email: imran at smartitengineering.com
Mobile: +880-1711402557
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From michel at extricom.com  Tue Mar  6 08:48:05 2007
From: michel at extricom.com (Michel Bensoussan)
Date: Tue, 06 Mar 2007 10:48:05 +0200
Subject: [SIPForum-discussion] REGISTER: Request or response?
Message-ID: <45ED2AC5.4010303@extricom.com>

Hello

Looking on RFC 3261, paragraph 20. Header Fields, Table 3, we can see 
that the Warning header may only appear in responses (r), and that it is 
optional (o) in the REGISTER (REG) method.

It this a contradiction?
REGISTER isn't by definition a Request and not a response?

Can we use Warning header in a REGISTER message?

If I cannot use Warning, is there a way to transmit a proprietary 
parameter in the REGISTER message?

Thanks.

Regards,
Michel.



From yahoosam at gmail.com  Tue Mar  6 14:43:33 2007
From: yahoosam at gmail.com (Sam Ernest Kumar Sam)
Date: Tue, 6 Mar 2007 09:43:33 -0500
Subject: [SIPForum-discussion] Sam Ernest Kumar has Tagged you! :)
Message-ID: <200703061443.l26EhX4r022368@sipforum.org>

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From Vinoth.E at mobax.com  Tue Mar  6 14:52:41 2007
From: Vinoth.E at mobax.com (vinoth)
Date: Tue,  6 Mar 2007 06:52:41 -0800
Subject: [SIPForum-discussion] query
Message-ID: <200703060652.AA107610166@mobax.com>

Hi.,

        If you havent Specified the Expires Header., it will Expires in 1 Hour by Default.
       
        Also you can Sent an Expires Header to some other duration such as 30 SECs.,etc locally in you SIP Message and you can Send it to the Server.


 Regards,
Vinoth Kumar.
Mobax Networks, Coimbatore.


---------- Original Message ----------------------------------
From: "Dawn Somen-NXDR43" <NXDR43 at motorola.com>
Date:  Thu, 1 Mar 2007 19:51:28 +0800

>HI!
> 
>Can someone tell me if no expiration is present, is it 3600sec by
>default or the server chooses a time interval for that as stated in sec
>10.2.1.1 of RFC3261?
>Also, is 360 configured locally?
> 
>Thanks!
>Regards,
>Somen
>
>
>


From Vinoth.E at mobax.com  Tue Mar  6 15:02:39 2007
From: Vinoth.E at mobax.com (vinoth)
Date: Tue,  6 Mar 2007 07:02:39 -0800
Subject: [SIPForum-discussion] About Header Contact
Message-ID: <200703060702.AA107348016@mobax.com>


Hi., 
    
          It can be anyone as you said., or even Both.,

For Eg:

Contact: sip:199.175.2.192:36000;

or

Contact: sip:euclid at parthenon.athens.gr

or

Contact: mailto:euclid at geometry.org

or (even two or More within the Same Message.)

Contact: sip:euclid at parthenon.athens.gr
Contact: mailto:euclid at geometry.org


Regards,
Vinoth Kumar.
Mobax Networks, Coimbatore.





---------- Original Message ----------------------------------
From: "Francesco Paolo Landolfo" <francesco.landolfo at gmail.com>
Date:  Thu, 1 Mar 2007 15:14:23 +0100

>Hi,
>I have the following doubt.
>Suppose that you have two client:
>
>   - A (sip:a at mydomain.org);
>   - B ( sip:b at mydomain.org).
>
>Suppose you have a Sip Proxy Server too.
>
>I want to setup a call session using Sip Proxy Server. (Please see the
>attached file)
>
>When B sends 180 RINGING back to the client A, does it writes in the Contact
>header its sip address  (Contact: "B" < sip:B at mydomain.org>) or its address
>of care (Contact: "A" < sip:B at 100.101.102.103>) ???
>
>What is the standard method?
>
>Thanks,
>Francesco Paolo Landolfo
>
>-- 
>Ci?? che facciamo in vita riecheggia nell'eternit??...(Il Gladiatore)
>"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
>presto." (C'era una volta in America)
>E adesso so cosa devo fare, devo continuare a respirare perch?? domani il
>sole sorger?? e chiss?? la marea cosa potr?? portare. (Cast Away)
>Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
>
>




From juanmoncel at yahoo.es  Wed Mar  7 08:24:20 2007
From: juanmoncel at yahoo.es (Juan Montero Celador)
Date: Wed, 7 Mar 2007 09:24:20 +0100
Subject: [SIPForum-discussion] Help with SIPp
Message-ID: <000d01c76092$03218a90$2802a8c0@int.satec.es>

Hello,

 

I am using SIPp for testing. I use it on Windows XP. I am having problems
for playing RTP: I have seen in the documentation the command "exec
play_pcap_audio" but it doesn't work, maybe because I am using WinPcap
instead of Pcap. Can anybody help me?

 

Regards,

--------------------------------

Juan Montero

 

"Mi infancia ha sido tan larga que nunca acaba de terminar"

 

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From lingzhi at xszhengda.com  Wed Mar  7 10:41:50 2007
From: lingzhi at xszhengda.com (lingzhi)
Date: Wed, 7 Mar 2007 18:41:50 +0800
Subject: [SIPForum-discussion] VPN sip/h323 voip gateway
Message-ID: <200703071841495898879@xszhengda.com>


Dear,

We are selling vpn voip gateway, 2 fxs & 4 fxs, both with voip client built in. 

VPN voip gateway is very useful in voip blocked area, and we already tested and well applied.

Plz feel free to contact if interested.

Looking forward to hearing from you.

Best regards, 
  
Ling
Marketing Director
+86 574 25713039-602
+86 13336888688, 13362487887
MSN: ling_zhi_ at hotmail.com
Yahoo: ling_zhi28
http://www.xszhengda.com
Sino-data Information Technology Ltd.




From wang.ran at byd.com.cn  Thu Mar  8 01:17:35 2007
From: wang.ran at byd.com.cn (wangran)
Date: Thu, 8 Mar 2007 09:17:35 +0800
Subject: [SIPForum-discussion] sip protocol questions.(two agent can't
	connect)
Message-ID: <OFA8A512A7.22C1B4C7-ON48257298.00072D67@byd.com.cn>

Dear all:

 

We have problem in sip calls, the attachment is capture the network packet.
Somebody who is familiar with sip protocol may help me analyse why the
problem came out.

 

The test environment is as this:

One sip client 192.168.1.26

 The other sip client      192.168.1.233

Ondo server(sip server) 192.168.1.13

 

 

Best of Regards,

 

wangran

 

 ***********************************************************************

BYD TECHFAITH?COMPANY?LIMITED(BTC)

 

Address:3/F,M8 West,NO.1 Jiu Xian Qiao Dong Road,

        Chao Yang District,Beijing,China  100016

PostCode:10016

Mobile: +86-13810362150

Tel: +86-10-58291226

Mail: wang.ran at byd.com.cn

 ***********************************************************************

 

 

 



Powered by BYD Security Gateway.

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From rtskarthik at gmail.com  Thu Mar  8 04:45:31 2007
From: rtskarthik at gmail.com (Karthik Arumugam)
Date: Thu, 8 Mar 2007 10:15:31 +0530
Subject: [SIPForum-discussion] Call transfer
Message-ID: <322cbb920703072045t42258398te880827ca0638f9c@mail.gmail.com>

*Hi All*

        *Scenario: User A on the soft phone makes a call to the PSTN
respondent*.*User A is in call with PSTN respondent. Now User A wants to
transfer call to the User C on the soft phone residing at same area as that
of User A. *
**
*Are there any issues pertaining to the above scenario? **Regarding call
charges, call transfer possibilities*

*Regards*
*Karthik.A*
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From adityaakumar at hotmail.com  Thu Mar  8 09:38:42 2007
From: adityaakumar at hotmail.com (aditya kumar)
Date: Thu, 08 Mar 2007 09:38:42 +0000
Subject: [SIPForum-discussion] Unable to establish a call
Message-ID: <BAY109-F99DB9C601642DCA80EFB8CE790@phx.gbl>

Hi,

I have successfully register the X-lite soft phone with Asterisk server,But 
unable to establish a call with other extension,When I am trying to call 
,got  503 response (service unavailable)
The Asterisk verision use :asterisk-1.2.15
Whenever I am calling to other extension,On asterisk server I have found the 
following messages.can u tell me where I was wrong If anybody hae prior 
experience of this scenario.
---------------------------------------------------
" 8 14:09:28 NOTICE[2981]: app_dial.c:1055 dial_exec_full: Unable to create 
channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/192.168.98.47-097dc098' status is 
'CHANUNAVAIL'"
--------------------------------------------------

Looking for advance comment

Thanks
Aditya

_________________________________________________________________
"Airtel Song Catcher. Get your Hello Tunes instantly" 
http://www.airtel.in/songcatcher/SONG_CATCHER.html



From ip.telephony at hotmail.com  Thu Mar  8 14:24:32 2007
From: ip.telephony at hotmail.com (mshari and abdulmalik KSU)
Date: Thu, 08 Mar 2007 17:24:32 +0300
Subject: [SIPForum-discussion] Sip softphone
Message-ID: <BAY18-F392E82AA3025FA13CC5D380790@phx.gbl>

i need softphone (open source)

also
the Features are:
- using SIP protocol
- for Pocket PC (( windows mobile ))
-support Cisco Call manger.

i need it ,

regards
malokey

_________________________________________________________________
Express yourself instantly with MSN Messenger! Download today it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/



From lucentdave at vip.sina.com  Fri Mar  9 09:03:40 2007
From: lucentdave at vip.sina.com (Jacky.Wang)
Date: Fri, 9 Mar 2007 17:03:40 +0800
Subject: [SIPForum-discussion] Sip softphone
References: <BAY18-F392E82AA3025FA13CC5D380790@phx.gbl>
Message-ID: <000d01c7622a$8d59a780$04edeedd@unix3g>

eyeBeam , a good soft sip-based terminal, maybe it is what you need.
you could down it from internet.


----- Original Message ----- 
From: "mshari and abdulmalik KSU" <ip.telephony at hotmail.com>
To: <discussion at sipforum.org>
Sent: Thursday, March 08, 2007 10:24 PM
Subject: [SIPForum-discussion] Sip softphone


>i need softphone (open source)
> 
> also
> the Features are:
> - using SIP protocol
> - for Pocket PC (( windows mobile ))
> -support Cisco Call manger.
> 
> i need it ,
> 
> regards
> malokey
> 
> _________________________________________________________________
> Express yourself instantly with MSN Messenger! Download today it's FREE! 
> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
> 
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>



From gurudattbalaji at gmail.com  Fri Mar  9 13:09:46 2007
From: gurudattbalaji at gmail.com (gurudatt balaji)
Date: Fri, 9 Mar 2007 18:39:46 +0530
Subject: [SIPForum-discussion] Query on IMS
Message-ID: <85ddb520703090509u1b160f65ob3e5a84efd7fe37a@mail.gmail.com>

Hi All,

With referrence to attachment, the scenario is wrt unregistered user.

When user is not registred in s-cscf, how i-cscf will find terminating
s-cscf and send invite to s-cscf.

HSS doesnot know the serving s-cscf because user is not registered.

In actual scenario, i-cscf fwds invite to s-cscf and then to voice mail (if
any)

Any idea......

Reg
Balaji
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From ravishankar.shiroor at wipro.com  Fri Mar  9 14:07:03 2007
From: ravishankar.shiroor at wipro.com (ravishankar.shiroor at wipro.com)
Date: Fri, 9 Mar 2007 19:37:03 +0530
Subject: [SIPForum-discussion] Query on IMS
In-Reply-To: <85ddb520703090509u1b160f65ob3e5a84efd7fe37a@mail.gmail.com>
References: <85ddb520703090509u1b160f65ob3e5a84efd7fe37a@mail.gmail.com>
Message-ID: <532B18E13CF9E64380EF5FDDE265E071C6E588@blr-m2-msg.wipro.com>

if the user has subscribed to services like voicemail/missed call
notifications (any service that is triggered when the user is not
registered),
the service (from an AS) will register on behalf of the user. it will
unregister when the user himself registers. the control details depend
ofcourse on how the service is designed.
 
regards,
ravi.
 
--
 
Ravishankar. G. Shiroor
Wipro Technologies, Bangalore.
 
ravishankar.shiroor at wipro.com
--
 


________________________________

	From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of gurudatt balaji
	Sent: Friday, March 09, 2007 6:40 PM
	To: discussion at sipforum.org
	Subject: [SIPForum-discussion] Query on IMS
	
	
	Hi All,
	 
	With referrence to attachment, the scenario is wrt unregistered
user.
	 
	When user is not registred in s-cscf, how i-cscf will find
terminating s-cscf and send invite to s-cscf.
	 
	HSS doesnot know the serving s-cscf because user is not
registered.
	 
	In actual scenario, i-cscf fwds invite to s-cscf and then to
voice mail (if any)
	 
	Any idea......
	 
	Reg
	Balaji

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From haloha201 at yahoo.com  Mon Mar 12 03:16:07 2007
From: haloha201 at yahoo.com (ha do)
Date: Sun, 11 Mar 2007 20:16:07 -0700 (PDT)
Subject: [SIPForum-discussion] Question on 1customer(Taxi service) want to
	have 10 calls at the same time
Message-ID: <397842.84239.qm@web32410.mail.mud.yahoo.com>

Hi 

i have trouble on my customer. The customer want to have 10 calls at the same time
i have Siemens product : HiQ 4200, HiQ 8000

Please give me some advise
Thanks

 
---------------------------------
 Get your own web address.
 Have a HUGE year through Yahoo! Small Business.
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From haloha201 at yahoo.com  Mon Mar 12 04:16:14 2007
From: haloha201 at yahoo.com (ha do)
Date: Sun, 11 Mar 2007 21:16:14 -0700 (PDT)
Subject: [SIPForum-discussion] question on recieving 10 calls at the same
	time on 1 phone number
Message-ID: <468435.97372.qm@web32411.mail.mud.yahoo.com>


Hi 

i have trouble on my customer. The customer want to have 10 calls at the same time on 1 phone number
i have Siemens product : HiQ 4200, HiQ 8000

Please give me some advise
Thanks
  

 
---------------------------------
Now that's room service! Choose from over 150,000 hotels 
in 45,000 destinations on Yahoo! Travel to find your fit.
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From hemantmehar at gmail.com  Mon Mar 12 05:56:15 2007
From: hemantmehar at gmail.com (hemant mehar)
Date: Mon, 12 Mar 2007 11:26:15 +0530
Subject: [SIPForum-discussion] what's the procedure for xlite registration???
Message-ID: <a26ae8fb0703112256p182de41i3c85931f1c947d37@mail.gmail.com>

Hi All,

Can anybody please tell me, what is the procedure to register xlite.

please write in detail???

thanks in advance
Hemant
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From zeroroot at tmax.co.kr  Mon Mar 12 09:12:16 2007
From: zeroroot at tmax.co.kr (Young-Geun Park)
Date: Mon, 12 Mar 2007 18:12:16 +0900
Subject: [SIPForum-discussion] according a performance tester for sipservlet
Message-ID: <200703120910.l2C9AsOu023986@sipforum.org>

Hi, all 

 

Is there any performance tester for sipservlet(JSR116)?

 

Regards, 

Park

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From gurudattbalaji at gmail.com  Mon Mar 12 12:23:43 2007
From: gurudattbalaji at gmail.com (gurudatt balaji)
Date: Mon, 12 Mar 2007 17:53:43 +0530
Subject: [SIPForum-discussion] Video Mail
Message-ID: <85ddb520703120523j6ed31d7eu4490ee1f3c07e4e1@mail.gmail.com>

Hi all,

Is there any standard/spec/RFCs for Video Mail.

Trying to integrate Video Mail with IMS.

Pls confirm

Reg
Balaji
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From chuwfan at gmail.com  Tue Mar 13 07:43:29 2007
From: chuwfan at gmail.com (Chuw Fan Lee)
Date: Tue, 13 Mar 2007 15:43:29 +0800
Subject: [SIPForum-discussion] [HELP]about UA to test SIPProxy
Message-ID: <8a5302dd0703130043u9a49f0bxe8e8a4678a2adc7d@mail.gmail.com>

i just have a sip proxy server here, (mjsip),
i try to connect with window messenger 5.1 with remote client,
it connected but the message can't forward to the other client from remote
client.
anyone can help me please?
urgent.
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From Shanmukharao.Makkapati at airtel.in  Tue Mar 13 11:21:51 2007
From: Shanmukharao.Makkapati at airtel.in (Shanmukharao.Makkapati at airtel.in)
Date: Tue, 13 Mar 2007 16:51:51 +0530
Subject: [SIPForum-discussion] Instant Messaging
Message-ID: <OFBACB3D6F.A6CE6106-ON6525729D.003E3E7B-6525729D.003DFBBF@bharti.com>


HI,
   Can any one help regarding the procedure involved in end to end instant 
messaging. Im just new to SIP. I want to know it in brief. can i get help 
from the forum.


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From rjsparks at nostrum.com  Tue Mar 13 15:05:27 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Tue, 13 Mar 2007 10:05:27 -0500
Subject: [SIPForum-discussion] SIPit 20 registration deadline is March 30
Message-ID: <EC28212F-39FF-49D8-BBD4-26C62F737623@nostrum.com>

If you plan to attend SIPit 20 in Antwerp, Belgium April 16-20
and have not yet registered, now's the time. Registration closes in
just over two weeks (March 30).

See www.sipit.net for more information and the registration link.

RjS


From Shanmukharao.Makkapati at airtel.in  Wed Mar 14 04:42:46 2007
From: Shanmukharao.Makkapati at airtel.in (Shanmukharao.Makkapati at airtel.in)
Date: Wed, 14 Mar 2007 10:12:46 +0530
Subject: [SIPForum-discussion] Modules involved in IM
Message-ID: <OFAC42D951.AACF88E2-ON6525729E.0019A397-6525729E.001971C9@bharti.com>


Hi,
  Can i get info about what the modules included in SIP IM project on C & 
Linux platform and what the structure of sdlc inlcuded, how it will be...?
please help me regard to this.



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From deepaknivas at rediffmail.com  Thu Mar 15 07:16:21 2007
From: deepaknivas at rediffmail.com (Deepak nivas)
Date: 15 Mar 2007 07:16:21 -0000
Subject: [SIPForum-discussion] query
Message-ID: <20070315071621.31672.qmail@webmail103.rediffmail.com>

  
Hi,


  In RFC 3261 chapter 17.1.2 deals with Non-INVITE Client Transaction.
for non-invite method proxy or no one will send provisional response.
what is need of Proceeding state. how come it will come from Trying state to Proceeding state if there is no provisonal response.


Thanks in advance.



Regards,
Deepak.
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From arb3 at alcatel-lucent.com  Thu Mar 15 15:56:37 2007
From: arb3 at alcatel-lucent.com (Bajracharya, Amar R (Amar))
Date: Thu, 15 Mar 2007 10:56:37 -0500
Subject: [SIPForum-discussion] SIP-I to SIP-I
In-Reply-To: <20070315071621.31672.qmail@webmail103.rediffmail.com>
References: <20070315071621.31672.qmail@webmail103.rediffmail.com>
Message-ID: <5C5FD7DED22D4D44AA7B9C853B70E06D9EC032@ILEXC1U02.ndc.lucent.com>

I have not seen any RFCs which talk about SIP-I(with ISUP encapsulation)
to SIP-I scenario. RFC3398 and ANSI T1.679 covers mostly on ISUP to
SIP-I and SIP-I to ISUP.

 

Does any body know how to handle the following scenario?

 

If you are acting as proxy and you get 183 with SDP and 183 with ACM
separately in 2 different messages, are you supposed to

 

-          Pass it along to as it is, OR

-          Drop 183 with SDP(as it doesn't contain any ISUP
encapsulation) , OR

-          Combine 2 and send 183 with SDP+ACM

 

Really appreciate it if anybody has any solution for this.

 

Thanks,

Amar Bajracharya

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From deepanshu at huawei.com  Fri Mar 16 07:26:42 2007
From: deepanshu at huawei.com (Deepanshu)
Date: Fri, 16 Mar 2007 15:26:42 +0800
Subject: [SIPForum-discussion] About ENUM
Message-ID: <014101c7679c$72251360$8178a40a@china.huawei.com>

Hi

ENUM query maps a Tel:URI (008613585147627) to a SIP-URI (sip:user at example.com). Can ENUM query work in the reverse direction mapping a SIP:URI to Tel:URI?

In other words i have a SIP-URI i like to know its corresponding Tel:URI, can i use ENUM query for the same.

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC
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From deepanshu at huawei.com  Fri Mar 16 08:40:33 2007
From: deepanshu at huawei.com (Deepanshu)
Date: Fri, 16 Mar 2007 16:40:33 +0800
Subject: [SIPForum-discussion] About ENUM
References: <E0C5A9C102C96D4989B9B4380A2AB1D002D6B859@ecnbjmw511.eapac.ericsson.se>
Message-ID: <017901c767a6$c30e12e0$8178a40a@china.huawei.com>

It is something related with SMS-IM interworking.

When an IM user sends a SIP MESSAGE (IMS domain) to a SMS-only user (CS
domain) then their is a need to find the Tel:URI associated with the SIP-URI
(included in the SIP MESSAGE) of the sender. The Tel:URI is needed to be
incorporated in the SMS message send towards receiver using which the
receiver can respond.

HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC


----- Original Message -----
From: "Rick Yang (GZ/CBC)" <rick.yang at ericsson.com>
To: "Deepanshu" <deepanshu at huawei.com>
Sent: Friday, March 16, 2007 4:24 PM
Subject: RE: [SIPForum-discussion] About ENUM


Hi,
May I ask what's the driven force behind this idea?


-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Deepanshu
Sent: 2007?3?16? 15:27
To: discussion at sipforum.org
Subject: [SIPForum-discussion] About ENUM

Hi

ENUM query maps a Tel:URI (008613585147627) to a SIP-URI
(sip:user at example.com). Can ENUM query work in the reverse direction mapping
a SIP:URI to Tel:URI?

In other words i have a SIP-URI i like to know its corresponding Tel:URI,
can i use ENUM query for the same.

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC




From deepanshu at huawei.com  Fri Mar 16 09:19:13 2007
From: deepanshu at huawei.com (Deepanshu)
Date: Fri, 16 Mar 2007 17:19:13 +0800
Subject: [SIPForum-discussion] About ENUM
References: <E0C5A9C102C96D4989B9B4380A2AB1D002D6B8FA@ecnbjmw511.eapac.ericsson.se>
Message-ID: <01b201c767ac$2a17b0e0$8178a40a@china.huawei.com>

Okey,

Thanks for you comments

But, one more issue. It is possible in IM that a user is using a PC or PDA
instead of mobile. In this case i believe he must not have a Tel:URI.
right??

So, how to do in this condition? any ideas

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC

----- Original Message -----
From: "Rick Yang (GZ/CBC)" <rick.yang at ericsson.com>
To: "Deepanshu" <deepanshu at huawei.com>
Sent: Friday, March 16, 2007 4:59 PM
Subject: RE: [SIPForum-discussion] About ENUM


I c.
During the registration, the SIP URI of sender would be registered as well
as the tel URL in the S-CSCF (Implicit registration).
So the tel URL is easy to get from the S-CSCF when it's necessary (IM case).

No ENUM should be concerned I belive.


-----Original Message-----
From: Deepanshu [mailto:deepanshu at huawei.com]
Sent: 2007?3?16? 16:41
To: Rick Yang (GZ/CBC)
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] About ENUM

It is something related with SMS-IM interworking.

When an IM user sends a SIP MESSAGE (IMS domain) to a SMS-only user (CS
domain) then their is a need to find the Tel:URI associated with the SIP-URI
(included in the SIP MESSAGE) of the sender. The Tel:URI is needed to be
incorporated in the SMS message send towards receiver using which the
receiver can respond.

HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC


----- Original Message -----
From: "Rick Yang (GZ/CBC)" <rick.yang at ericsson.com>
To: "Deepanshu" <deepanshu at huawei.com>
Sent: Friday, March 16, 2007 4:24 PM
Subject: RE: [SIPForum-discussion] About ENUM


Hi,
May I ask what's the driven force behind this idea?


-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Deepanshu
Sent: 2007?3?16? 15:27
To: discussion at sipforum.org
Subject: [SIPForum-discussion] About ENUM

Hi

ENUM query maps a Tel:URI (008613585147627) to a SIP-URI
(sip:user at example.com). Can ENUM query work in the reverse direction mapping
a SIP:URI to Tel:URI?

In other words i have a SIP-URI i like to know its corresponding Tel:URI,
can i use ENUM query for the same.

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC






From francesco.landolfo at gmail.com  Fri Mar 16 10:29:57 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Fri, 16 Mar 2007 11:29:57 +0100
Subject: [SIPForum-discussion] SIP - Instant Messaging with Session
Message-ID: <d08f948c0703160329r48298459ma99634992d49d2ab@mail.gmail.com>

Hi,
I have read about the possibility to make IM with SIP session.
I'd like to have some other information about this solution and in
particular about a Many to Many chat with Session.

Please can someone give me some links, papers, RFCs... in which I can learn
this things?

If someone does not understand what I mean, please contact me.

Thanks in avoidance,
Francesco Paolo Landolfo

-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
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From erolturac at gmail.com  Fri Mar 16 13:47:07 2007
From: erolturac at gmail.com (erol turac)
Date: Fri, 16 Mar 2007 15:47:07 +0200
Subject: [SIPForum-discussion] SIP - Instant Messaging with Session
In-Reply-To: <d08f948c0703160329r48298459ma99634992d49d2ab@mail.gmail.com>
References: <d08f948c0703160329r48298459ma99634992d49d2ab@mail.gmail.com>
Message-ID: <3079c79a0703160647v6e01a361u6df6b4c9722f647d@mail.gmail.com>

IM is an extension for sip and rfc3428 describes how it should be
implemented.


On 3/16/07, Francesco Paolo Landolfo <francesco.landolfo at gmail.com> wrote:
>
> Hi,
> I have read about the possibility to make IM with SIP session.
> I'd like to have some other information about this solution and in
> particular about a Many to Many chat with Session.
>
> Please can someone give me some links, papers, RFCs... in which I can
> learn this things?
>
> If someone does not understand what I mean, please contact me.
>
> Thanks in avoidance,
> Francesco Paolo Landolfo
>
> --
> Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
> "Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
> presto." (C'era una volta in America)
> E adesso so cosa devo fare, devo continuare a respirare perch? domani il
> sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
> Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>


-- 
Erol Tura?
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From francesco.landolfo at gmail.com  Fri Mar 16 13:55:04 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Fri, 16 Mar 2007 14:55:04 +0100
Subject: [SIPForum-discussion] SIP - Instant Messaging with Session
In-Reply-To: <3079c79a0703160647v6e01a361u6df6b4c9722f647d@mail.gmail.com>
References: <d08f948c0703160329r48298459ma99634992d49d2ab@mail.gmail.com>
	<3079c79a0703160647v6e01a361u6df6b4c9722f647d@mail.gmail.com>
Message-ID: <d08f948c0703160655x33450351q22a4d7a63ea8a0a8@mail.gmail.com>

Yes, I know what you say but RFC 3428 does not describe in details the IM
chat using Session that is what I try.

Thanks in anticipation,
Francesco


Il 16/03/07, erol turac <erolturac at gmail.com> ha scritto:
>
> IM is an extension for sip and rfc3428 describes how it should be
> implemented.
>
>
> On 3/16/07, Francesco Paolo Landolfo < francesco.landolfo at gmail.com>
> wrote:
>
> > Hi,
> > I have read about the possibility to make IM with SIP session.
> > I'd like to have some other information about this solution and in
> > particular about a Many to Many chat with Session.
> >
> > Please can someone give me some links, papers, RFCs... in which I can
> > learn this things?
> >
> > If someone does not understand what I mean, please contact me.
> >
> > Thanks in avoidance,
> > Francesco Paolo Landolfo
> >
> > --
> > Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
> > "Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
> > presto." (C'era una volta in America)
> > E adesso so cosa devo fare, devo continuare a respirare perch? domani il
> > sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
> > Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
> > _______________________________________________
> > This is the SIP Forum discussion mailing list
> > TO UNSUBSCRIBE, or edit your delivery options, please visit
> > http://sipforum.org/mailman/listinfo/discussion
> > Post to the list at discussion at sipforum.org
> >
> >
>
>
> --
> Erol Tura?




-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
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From pavan.jain at ingenient.com  Fri Mar 16 13:58:04 2007
From: pavan.jain at ingenient.com (Pavan Jain)
Date: Fri, 16 Mar 2007 09:58:04 -0400
Subject: [SIPForum-discussion] List of registered user on SIP server
Message-ID: <EXCHANGEunLOyDc41W200000497@mail.ingenient.com>

Hello All:

 

How can I get the list of registered users on the SIP server?

 

Regards,

Pavan

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From wellya at wellya.net  Sat Mar 17 06:24:45 2007
From: wellya at wellya.net (wellya)
Date: Sat, 17 Mar 2007 14:24:45 +0800
Subject: [SIPForum-discussion] List of registered user on SIP server
References: <EXCHANGEunLOyDc41W200000497@mail.ingenient.com>
Message-ID: <200703171424417035768@wellya.net>

you can query database




wellya
2007-03-17



???? Pavan Jain
????? 2007-03-16 22:20:43
???? discussion at sipforum.org
??? 
??? [SIPForum-discussion] List of registered user on SIP server

Hello All:
 
How can I get the list of registered users on the SIP server?
 
Regards,
Pavan
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From wellya at wellya.net  Sat Mar 17 06:26:19 2007
From: wellya at wellya.net (wellya)
Date: Sat, 17 Mar 2007 14:26:19 +0800
Subject: [SIPForum-discussion] SIP - Instant Messaging with Session
References: <d08f948c0703160329r48298459ma99634992d49d2ab@mail.gmail.com>
	<3079c79a0703160647v6e01a361u6df6b4c9722f647d@mail.gmail.com>
	<d08f948c0703160655x33450351q22a4d7a63ea8a0a8@mail.gmail.com>
Message-ID: <200703171426185628520@wellya.net>

Maybe you can search it in http://www.tech-invite.com/ or http://www.rfc-editor.org and http://www.wellya.net 





wellya
2007-03-17



???? Francesco Paolo Landolfo
????? 2007-03-16 22:17:37
???? erol turac
??? discussion at sipforum.org
??? Re: [SIPForum-discussion] SIP - Instant Messaging with Session

Yes, I know what you say but RFC 3428 does not describe in details the IM chat using Session that is what I try.

Thanks in anticipation,
Francesco



Il 16/03/07, erol turac <erolturac at gmail.com> ha scritto:
IM is an extension for sip and rfc3428 describes how it should be implemented. 



On 3/16/07, Francesco Paolo Landolfo < francesco.landolfo at gmail.com> wrote:
Hi,
I have read about the possibility to make IM with SIP session.
I'd like to have some other information about this solution and in particular about a Many to Many chat with Session.

Please can someone give me some links, papers, RFCs... in which I can learn this things? 

If someone does not understand what I mean, please contact me.

Thanks in avoidance,
Francesco Paolo Landolfo

-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore) 
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il sole sorger? e chiss? la marea cosa potr? portare. (Cast Away) 
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso) 

_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org





-- 
Erol Tura? 



-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto presto." (C'era una volta in America) 
E adesso so cosa devo fare, devo continuare a respirare perch? domani il sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso) 
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From danishzaidi54 at yahoo.com  Sat Mar 17 16:33:18 2007
From: danishzaidi54 at yahoo.com (Danish Zaidi)
Date: Sat, 17 Mar 2007 09:33:18 -0700 (PDT)
Subject: [SIPForum-discussion] can any 1 give me samples for CallHold for
	Jain-Sip API
In-Reply-To: <200703171424417035768@wellya.net>
Message-ID: <265790.55020.qm@web90611.mail.mud.yahoo.com>

Can someone give me sample codes for CallHold in JainSip API..

thanx in advance

 
---------------------------------
Food fight? Enjoy some healthy debate
in the Yahoo! Answers Food & Drink Q&A.
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From carol_chang8009 at yahoo.com.cn  Mon Mar 19 06:14:06 2007
From: carol_chang8009 at yahoo.com.cn (Carol Chang)
Date: Mon, 19 Mar 2007 14:14:06 +0800 (CST)
Subject: [SIPForum-discussion] YUXIN news about new model launched
In-Reply-To: <200703171426185628520@wellya.net>
Message-ID: <93030.86408.qm@web15804.mail.cnb.yahoo.com>

   
   YUXIN NEW MODEL YWH201,YWH202 PHONES LAUNCHED
        Through development and experiment for about one year,Zhengzhou Yuneng Communication Co.,Ltd launched a new model YWH202 IP phone,which base on Infineon chip and support SIP,IAX2,H323 protocols,build-in-Router,3-way conference calls,voice message,recording,two SIP servers running at the same time.The highly voice effect ,the friendly menu operation, the easy setting page,and the low fees which can make you enjoy high tech product and save phone expense without too much specialized knowledge.
  At the same time, YWH201 IP Phone based on AR1688 chip and supporting SIP,IAX2 protocol has seized hold of the market by sale of several ten thousands units since it had been released on Oct.2006.This model of phone was developed together by outstanding workers from Aredfox and Zhengzhou Yuneng ,the former is famous in suppling solution in international VOIP field,and the latter has six years excellent performance in sale of internet phones.The type of YWH201 not only inherits merites of the phones based on PA1688,but also has many new functions(side sound,voice message and so on).
  As a researcher and producer in domestic VOIP terminal device,Zhengzhou Yueng communication Co.,Ltd has complete production lines and quality control systems,import and export license,CE and FCC certificate, and also devotes to new technical development and application.It watches colsely the development of VOIP market and the demand of customers ,meanwhile release new products and new functions to make customers satisfied.At present we have YWH100,YWH10,YWH200,YWH300,YWH500(POE)series products based on PA1688 chip,and our YWH200C,YWH300C,YWH600A,YWH600B(True FXO port) support 3-way conference calls,build-in-Router,registering on 3 SIP servers at the same time(switching freely).In future we will release more and more phones which based on AR1688 and Infineon chips.Regarding to gateways,we have had YGW20 based onPA1688,and YGW30A,YGW30B(true FXO port) based on CM5000 chip. These products has been validated and accepted by market.We  cooperate with many companies engaged in
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  At present the ATA YGW50(1FXS0),YGW60(2FXS) which based on Infineon chip has passed testing already and will be put into market soonly.
  We will try our best to provide high quality products and multi-position services for VOIP friends,and make YUXIN first-class brand in the domestic VOIP terminal device! 
   
  -------------------------------------------
ZHENNGZHOU YUNENG COMMUNICATION CO.,LTD
1KANGLELI,ZHONGYUAN DISTRICT,ZHENGZHOU,PRC
URL:HTTP://WWW.YNTX.COM
TEL:86-371-67657240
FAX:86-371-67657239
MSN:Carol_chang8009 at hotmail.com

   
   
   
   
   
   
   


 		
---------------------------------
????????-3.5G???20M??? 
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From mweiglh at ist.tugraz.at  Mon Mar 19 13:01:25 2007
From: mweiglh at ist.tugraz.at (Martin Weiglhofer)
Date: Mon, 19 Mar 2007 14:01:25 +0100
Subject: [SIPForum-discussion] Proxy-to-User Authentication
Message-ID: <45FE89A5.5060704@ist.tugraz.at>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Dear all,

assume, that a user tries to register a contact address on a registrar
which requires authentication, but the user is not known by the
registrar. The user sends a REGISTER message (without authentication
credentials) which is answered with a "401 Unauthorized" response. Now
the user sends a second REGISTER request which includes an Authorization
header field. What should be the response of the server to the second
REGISTER request? Should the registrar again reply with a "401
Unauthorized", or should the registrar reject the request with a "404
Not Found"?

Thanks in advance.

Regards
Martin

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Version: GnuPG v1.4.6 (GNU/Linux)
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From mweiglh at ist.tugraz.at  Mon Mar 19 13:01:37 2007
From: mweiglh at ist.tugraz.at (Martin Weiglhofer)
Date: Mon, 19 Mar 2007 14:01:37 +0100
Subject: [SIPForum-discussion] Proxy-to-User Authentication
Message-ID: <45FE89B1.8020103@ist.tugraz.at>

Dear all,

assume, that a user tries to register a contact address on a registrar
which requires authentication, but the user is not known by the
registrar. The user sends a REGISTER message (without authentication
credentials) which is answered with a "401 Unauthorized" response. Now
the user sends a second REGISTER request which includes an Authorization
header field. What should be the response of the server to the second
REGISTER request? Should the registrar again reply with a "401
Unauthorized", or should the registrar reject the request with a "404
Not Found"?

Thanks in advance.

Regards
Martin



From rjsparks at nostrum.com  Mon Mar 19 13:35:04 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Mon, 19 Mar 2007 14:35:04 +0100
Subject: [SIPForum-discussion] Proxy-to-User Authentication
In-Reply-To: <45FE89A5.5060704@ist.tugraz.at>
References: <45FE89A5.5060704@ist.tugraz.at>
Message-ID: <7D2E14F7-096F-4246-97B0-392A0E6EEC25@nostrum.com>

It would be an odd edge condition for it to be appropriate for you to  
return
a 404 (given that you returned a 401 the first time). That means that  
whoever
resubmitted the request with credentials has credentials that you are  
willing
to accept as valid for a resource you don't know about. If you have a  
policy
that anyone with an account can modify the registration for any AoR  
on your
system, I could see this happening. Typical systems bind a set of  
credentials
fairly tightly to an AoR (this username password can only be used  
with this AoR
and its the only username password pair I'll accept for that AoR). In  
that situation,
returning a 404 would only make sense if you had recently deleted the  
resource
but hadn't invalidated the credentials yet.

If the credentials in the second request aren't good, you'll return  
another 401.

RjS

On Mar 19, 2007, at 2:01 PM, Martin Weiglhofer wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Dear all,
>
> assume, that a user tries to register a contact address on a registrar
> which requires authentication, but the user is not known by the
> registrar. The user sends a REGISTER message (without authentication
> credentials) which is answered with a "401 Unauthorized" response. Now
> the user sends a second REGISTER request which includes an  
> Authorization
> header field. What should be the response of the server to the second
> REGISTER request? Should the registrar again reply with a "401
> Unauthorized", or should the registrar reject the request with a "404
> Not Found"?
>
> Thanks in advance.
>
> Regards
> Martin
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.6 (GNU/Linux)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFF/oml6XHVH58yroMRAuH7AKCHsGhkTlLznZKVpLw93FXAqnKIuwCgwCMy
> E06PmQIXSxEV62ajCjmDqEk=
> =UvKG
> -----END PGP SIGNATURE-----
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit http:// 
> sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org



From pavan.jain at ingenient.com  Mon Mar 19 14:23:56 2007
From: pavan.jain at ingenient.com (Pavan Jain)
Date: Mon, 19 Mar 2007 10:23:56 -0400
Subject: [SIPForum-discussion] List of registered user on SIP server
In-Reply-To: <200703171424417035768@wellya.net>
Message-ID: <EXCHANGEtpIuPHDXE2s00000589@mail.ingenient.com>

Is there a message format that I need to send to Server( similar to REGISTER)? An example will be helpful.

 

Regards,

Pavan

 

  _____  

From: wellya [mailto:wellya at wellya.net] 
Sent: Saturday, March 17, 2007 2:25 AM
To: Pavan Jain; discussion at sipforum.org
Subject: Re: [SIPForum-discussion] List of registered user on SIP server

 

you can query database

 

  _____  

wellya

2007-03-17

  _____  

???? Pavan Jain

????? 2007-03-16 22:20:43

???? discussion at sipforum.org

??? 

??? [SIPForum-discussion] List of registered user on SIP server

 

Hello All:

 

How can I get the list of registered users on the SIP server?

 

Regards,

Pavan

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From mweiglh at ist.tugraz.at  Mon Mar 19 14:52:22 2007
From: mweiglh at ist.tugraz.at (Martin Weiglhofer)
Date: Mon, 19 Mar 2007 15:52:22 +0100
Subject: [SIPForum-discussion] Proxy-to-User Authentication
In-Reply-To: <7D2E14F7-096F-4246-97B0-392A0E6EEC25@nostrum.com>
References: <45FE89A5.5060704@ist.tugraz.at>
	<7D2E14F7-096F-4246-97B0-392A0E6EEC25@nostrum.com>
Message-ID: <45FEA3A6.1030600@ist.tugraz.at>

Thanks for the fast answer.

If I understood your explanation correctly, that means that I might
mislead the user, such that the user thinks password or username for the
authentication are incorrect. Instead the user might have configured the
wrong registration server on the SIP phone.


Robert Sparks wrote:
> It would be an odd edge condition for it to be appropriate for you to
> return
> a 404 (given that you returned a 401 the first time). That means that
> whoever
> resubmitted the request with credentials has credentials that you are
> willing
> to accept as valid for a resource you don't know about. If you have a
> policy
> that anyone with an account can modify the registration for any AoR on your
> system, I could see this happening. Typical systems bind a set of
> credentials
> fairly tightly to an AoR (this username password can only be used with
> this AoR
> and its the only username password pair I'll accept for that AoR). In
> that situation,
> returning a 404 would only make sense if you had recently deleted the
> resource
> but hadn't invalidated the credentials yet.
> 
> If the credentials in the second request aren't good, you'll return
> another 401.
> 
> RjS
> 
> On Mar 19, 2007, at 2:01 PM, Martin Weiglhofer wrote:
> 
> Dear all,
> 
> assume, that a user tries to register a contact address on a registrar
> which requires authentication, but the user is not known by the
> registrar. The user sends a REGISTER message (without authentication
> credentials) which is answered with a "401 Unauthorized" response. Now
> the user sends a second REGISTER request which includes an Authorization
> header field. What should be the response of the server to the second
> REGISTER request? Should the registrar again reply with a "401
> Unauthorized", or should the registrar reject the request with a "404
> Not Found"?
> 
> Thanks in advance.
> 
> Regards
> Martin
> 
_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org

-- 
Martin Weiglhofer
Inst. f. Software Technology - Graz University of Technology
Inffeldgasse 16b/II - 8010 Graz - Austria
phone: ++43 316 873 5763
mail: weiglhofer at ist.tugraz.at
web: http://www.ist.tugraz.at/staff/weiglhofer


From danishzaidi54 at yahoo.com  Mon Mar 19 17:24:21 2007
From: danishzaidi54 at yahoo.com (Danish Zaidi)
Date: Mon, 19 Mar 2007 10:24:21 -0700 (PDT)
Subject: [SIPForum-discussion] Need Help with SIP Call Hold
Message-ID: <282483.87232.qm@web90605.mail.mud.yahoo.com>

Dear All

m having trouble with SIP Call Hold in Jain SIP,
didn't find any sample code for Call Hold.

it will be very much helpful for me if any 1 gives me
sample for Call Hold in Jain Sip.


thanx in Advance
Danish Zaidi


 
____________________________________________________________________________________
Don't pick lemons.
See all the new 2007 cars at Yahoo! Autos.
http://autos.yahoo.com/new_cars.html 


From francesco.landolfo at gmail.com  Tue Mar 20 10:23:53 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Tue, 20 Mar 2007 11:23:53 +0100
Subject: [SIPForum-discussion] List of registered user on SIP server
In-Reply-To: <EXCHANGEtpIuPHDXE2s00000589@mail.ingenient.com>
References: <200703171424417035768@wellya.net>
	<EXCHANGEtpIuPHDXE2s00000589@mail.ingenient.com>
Message-ID: <d08f948c0703200323o9352586xf3d42bf05ddad9d9@mail.gmail.com>

I think no.
Why would you try this message? In what scenario would you use it?

2007/3/19, Pavan Jain <pavan.jain at ingenient.com>:
>
>  Is there a message format that I need to send to Server( similar to
> REGISTER)? An example will be helpful.
>
>
>
> Regards,
>
> Pavan
>
>
>  ------------------------------
>
> *From:* wellya [mailto:wellya at wellya.net]
> *Sent:* Saturday, March 17, 2007 2:25 AM
> *To:* Pavan Jain; discussion at sipforum.org
> *Subject:* Re: [SIPForum-discussion] List of registered user on SIP server
>
>
>
> you can query database
>
>
>  ------------------------------
>
> wellya
>
> 2007-03-17
>   ------------------------------
>
> *????* Pavan Jain
>
> *?????* 2007-03-16 22:20:43
>
> *????* discussion at sipforum.org
>
> *???*
>
> *???* [SIPForum-discussion] List of registered user on SIP server
>
>
>
> Hello All:
>
>
>
> How can I get the list of registered users on the SIP server?
>
>
>
> Regards,
>
> Pavan
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>


-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
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From dinis_7 at hotmail.com  Tue Mar 20 10:32:11 2007
From: dinis_7 at hotmail.com (Nelson Dinis)
Date: Tue, 20 Mar 2007 10:32:11 +0000
Subject: [SIPForum-discussion] Types of sessions
Message-ID: <BAY112-F9758F0517CF80D228D832DE750@phx.gbl>

hello all,

I want to establish a session, using SIP, but in my case i don't want that 
the caller (person that send the INVITE) knows what will be the session, 
that need to be decide by the called (person that receive the Invite).

I was tinking sending multiple informacion (multipart body) on the INVITE, 
and then de called decide what will be the session. It possibile to do 
something like this?

It possible use SIP protocol to establish a session that is not media (voip, 
multimedia), for exemple i want to establish a VNC (Virtual Network 
Computing)?

Regards

Nelson Dinis
Portugal

_________________________________________________________________
MSN Hotmail, o maior webmail do Brasil. http://www.hotmail.com



From zeroroot at tmax.co.kr  Tue Mar 20 11:22:18 2007
From: zeroroot at tmax.co.kr (Young-Geun Park)
Date: Tue, 20 Mar 2007 20:22:18 +0900
Subject: [SIPForum-discussion]  how to compute cps(call per second)?
Message-ID: <200703201120.l2KBKelL006065@sipforum.org>

Hi, all

 

I want to know how to compute CPS(call per second) specially with a proxyApp
as follows.

 

UAC          Proxy         UAS

   1(invite)

 ------------>

   2(100 trying)

<-----------

   3(invite)

------------>

4(180 Ringing)

<-----------

5(180 Ringing)

<-----------

6(200 OK)

<-----------

7(200 OK)

<-----------

8(ACK)

------------>

   9(ACK)

------------>

10(BYE)

<-----------

11(BYE)

<-----------

12(200 OK)

------------>

   13(200 OK)

------------>

 

Thanks in advance

 

Regards, 

Park

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From zeroroot at tmax.co.kr  Tue Mar 20 11:31:43 2007
From: zeroroot at tmax.co.kr (Young-Geun Park)
Date: Tue, 20 Mar 2007 20:31:43 +0900
Subject: [SIPForum-discussion] how to compute cps(call per second)?
Message-ID: <200703201130.l2KBU3QY014175@sipforum.org>

and what are there performance factors related to a sip servlet container
that deploys the proxy app?

 

Park

  _____  

From: Young-Geun Park [mailto:zeroroot at tmax.co.kr] 
Sent: Tuesday, March 20, 2007 8:22 PM
To: 'discussion at sipforum.org'
Subject: [SIPForum-discussion] how to compute cps(call per second)?

 

Hi, all

 

I want to know how to compute CPS(call per second) specially with a proxyApp
as follows.

 

UAC          Proxy         UAS

   1(invite)

 ------------>

   2(100 trying)

<-----------

   3(invite)

------------>

4(180 Ringing)

<-----------

5(180 Ringing)

<-----------

6(200 OK)

<-----------

7(200 OK)

<-----------

8(ACK)

------------>

   9(ACK)

------------>

10(BYE)

<-----------

11(BYE)

<-----------

12(200 OK)

------------>

   13(200 OK)

------------>

 

Thanks in advance

 

Regards, 

Park

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From Shanmukharao.Makkapati at airtel.in  Tue Mar 20 13:37:00 2007
From: Shanmukharao.Makkapati at airtel.in (Shanmukharao.Makkapati at airtel.in)
Date: Tue, 20 Mar 2007 19:07:00 +0530
Subject: [SIPForum-discussion] Instant Messaging
Message-ID: <OFFAFFD5B2.88820935-ON652572A4.004A194F-652572A4.004A569E@bharti.com>


Hi,
  I want to get info about a breif sdlc architecture involved in Instant 
messenger using SIP on C, Linux for the developement of whole project.

Can any one help me regard 2 this.



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From pavan.jain at ingenient.com  Tue Mar 20 14:03:49 2007
From: pavan.jain at ingenient.com (Pavan Jain)
Date: Tue, 20 Mar 2007 10:03:49 -0400
Subject: [SIPForum-discussion] List of registered user on SIP server
In-Reply-To: <d08f948c0703200323o9352586xf3d42bf05ddad9d9@mail.gmail.com>
Message-ID: <EXCHANGEy7BXx4XFKpR000005f6@mail.ingenient.com>

Our product needs to know how many users are online (registered) just like we have in IM chat sessions. For that I was wondering my SIP stack can send a Query message from stack to Server, and get a list of registered users. 

 

  _____  

From: Francesco Paolo Landolfo [mailto:francesco.landolfo at gmail.com] 
Sent: Tuesday, March 20, 2007 6:24 AM
To: Pavan Jain
Cc: wellya; discussion at sipforum.org
Subject: Re: [SIPForum-discussion] List of registered user on SIP server

 

I think no. 
Why would you try this message? In what scenario would you use it?

2007/3/19, Pavan Jain <pavan.jain at ingenient.com>: 

Is there a message format that I need to send to Server( similar to REGISTER)? An example will be helpful.

 

Regards,

Pavan

 

  _____  

From: wellya [mailto:wellya at wellya.net] 
Sent: Saturday, March 17, 2007 2:25 AM
To: Pavan Jain; discussion at sipforum.org
Subject: Re: [SIPForum-discussion] List of registered user on SIP server

 

you can query database

  

  _____  

wellya 

2007-03-17 

  _____  

???? Pavan Jain

????? 2007-03-16 22:20:43

???? discussion at sipforum.org 

??? 

??? [SIPForum-discussion] List of registered user on SIP server

  

Hello All:

 

How can I get the list of registered users on the SIP server?

 

Regards,

Pavan


_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org




-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto presto." (C'era una volta in America) 
E adesso so cosa devo fare, devo continuare a respirare perch? domani il sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso) 

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From francesco.landolfo at gmail.com  Tue Mar 20 14:14:46 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Tue, 20 Mar 2007 15:14:46 +0100
Subject: [SIPForum-discussion] List of registered user on SIP server
In-Reply-To: <EXCHANGEy7BXx4XFKpR000005f6@mail.ingenient.com>
References: <d08f948c0703200323o9352586xf3d42bf05ddad9d9@mail.gmail.com>
	<EXCHANGEy7BXx4XFKpR000005f6@mail.ingenient.com>
Message-ID: <d08f948c0703200714u7cc8db99qfbaf90512494574c@mail.gmail.com>

I don't know how help you. I'm sorry.

Good luck,
Francesco

2007/3/20, Pavan Jain <pavan.jain at ingenient.com>:
>
>  Our product needs to know how many users are online (registered) just
> like we have in IM chat sessions. For that I was wondering my SIP stack can
> send a Query message from stack to Server, and get a list of registered
> users.
>
>
>  ------------------------------
>
> *From:* Francesco Paolo Landolfo [mailto:francesco.landolfo at gmail.com]
> *Sent:* Tuesday, March 20, 2007 6:24 AM
> *To:* Pavan Jain
> *Cc:* wellya; discussion at sipforum.org
> *Subject:* Re: [SIPForum-discussion] List of registered user on SIP server
>
>
>
> I think no.
> Why would you try this message? In what scenario would you use it?
>
> 2007/3/19, Pavan Jain <pavan.jain at ingenient.com>:
>
> Is there a message format that I need to send to Server( similar to
> REGISTER)? An example will be helpful.
>
>
>
> Regards,
>
> Pavan
>
>
>  ------------------------------
>
> *From:* wellya [mailto:wellya at wellya.net]
> *Sent:* Saturday, March 17, 2007 2:25 AM
> *To:* Pavan Jain; discussion at sipforum.org
> *Subject:* Re: [SIPForum-discussion] List of registered user on SIP server
>
>
>
> you can query database
>
>
>  ------------------------------
>
> wellya
>
> 2007-03-17
>   ------------------------------
>
> *????* Pavan Jain
>
> *?????* 2007-03-16 22:20:43
>
> *????* discussion at sipforum.org
>
> *???*
>
> *???* [SIPForum-discussion] List of registered user on SIP server
>
>
>
> Hello All:
>
>
>
> How can I get the list of registered users on the SIP server?
>
>
>
> Regards,
>
> Pavan
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
>
>
> --
> Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
> "Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
> presto." (C'era una volta in America)
> E adesso so cosa devo fare, devo continuare a respirare perch? domani il
> sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
> Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
>



-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
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From durgani at gmail.com  Tue Mar 20 14:51:58 2007
From: durgani at gmail.com (Prakash Durgani)
Date: Tue, 20 Mar 2007 10:51:58 -0400
Subject: [SIPForum-discussion] Types of sessions
In-Reply-To: <BAY112-F9758F0517CF80D228D832DE750@phx.gbl>
References: <BAY112-F9758F0517CF80D228D832DE750@phx.gbl>
Message-ID: <f6f514070703200751v3530f260s33f689b939d62cf9@mail.gmail.com>

Refer to RFC 3264 on Offer/Answer Model.

On 3/20/07, Nelson Dinis <dinis_7 at hotmail.com> wrote:
>
> hello all,
>
> I want to establish a session, using SIP, but in my case i don't want that
> the caller (person that send the INVITE) knows what will be the session,
> that need to be decide by the called (person that receive the Invite).
>
> I was tinking sending multiple informacion (multipart body) on the INVITE,
> and then de called decide what will be the session. It possibile to do
> something like this?
>
> It possible use SIP protocol to establish a session that is not media
> (voip,
> multimedia), for exemple i want to establish a VNC (Virtual Network
> Computing)?
>
> Regards
>
> Nelson Dinis
> Portugal
>
> _________________________________________________________________
> MSN Hotmail, o maior webmail do Brasil. http://www.hotmail.com
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
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From muraliwind at yahoo.com  Tue Mar 20 17:28:39 2007
From: muraliwind at yahoo.com (Murali ~~)
Date: Tue, 20 Mar 2007 10:28:39 -0700 (PDT)
Subject: [SIPForum-discussion] List of registered user on SIP server
In-Reply-To: <d08f948c0703200714u7cc8db99qfbaf90512494574c@mail.gmail.com>
Message-ID: <874838.94380.qm@web31903.mail.mud.yahoo.com>


Use SUBSCRIBE/NOTIFY!
http://www.ietf.org/rfc/rfc3265.txt

Cheers,
Murali


--- Francesco Paolo Landolfo
<francesco.landolfo at gmail.com> wrote:

> I don't know how help you. I'm sorry.
> 
> Good luck,
> Francesco
> 
> 2007/3/20, Pavan Jain <pavan.jain at ingenient.com>:
> >
> >  Our product needs to know how many users are
> online (registered) just
> > like we have in IM chat sessions. For that I was
> wondering my SIP stack can
> > send a Query message from stack to Server, and get
> a list of registered
> > users.
> >
> >
> >  ------------------------------
> >
> > *From:* Francesco Paolo Landolfo
> [mailto:francesco.landolfo at gmail.com]
> > *Sent:* Tuesday, March 20, 2007 6:24 AM
> > *To:* Pavan Jain
> > *Cc:* wellya; discussion at sipforum.org
> > *Subject:* Re: [SIPForum-discussion] List of
> registered user on SIP server
> >
> >
> >
> > I think no.
> > Why would you try this message? In what scenario
> would you use it?
> >
> > 2007/3/19, Pavan Jain <pavan.jain at ingenient.com>:
> >
> > Is there a message format that I need to send to
> Server( similar to
> > REGISTER)? An example will be helpful.
> >
> >
> >
> > Regards,
> >
> > Pavan
> >
> >
> >  ------------------------------
> >
> > *From:* wellya [mailto:wellya at wellya.net]
> > *Sent:* Saturday, March 17, 2007 2:25 AM
> > *To:* Pavan Jain; discussion at sipforum.org
> > *Subject:* Re: [SIPForum-discussion] List of
> registered user on SIP server
> >
> >
> >
> > you can query database
> >
> >
> >  ------------------------------
> >
> > wellya
> >
> > 2007-03-17
> >   ------------------------------
> >
> > *????????????* Pavan Jain
> >
> > *???????????????* 2007-03-16 22:20:43
> >
> > *????????????* discussion at sipforum.org
> >
> > *?????????*
> >
> > *?????????* [SIPForum-discussion] List of
> registered user on SIP server
> >
> >
> >
> > Hello All:
> >
> >
> >
> > How can I get the list of registered users on the
> SIP server?
> >
> >
> >
> > Regards,
> >
> > Pavan
> >
> >
> > _______________________________________________
> > This is the SIP Forum discussion mailing list
> > TO UNSUBSCRIBE, or edit your delivery options,
> please visit
> > http://sipforum.org/mailman/listinfo/discussion
> > Post to the list at discussion at sipforum.org
> >
> >
> >
> >
> > --
> > Ci?? che facciamo in vita riecheggia
> nell'eternit??...(Il Gladiatore)
> > "Noodles, cos'hai fatto in tutti questi anni?" "
> Sono andato a letto
> > presto." (C'era una volta in America)
> > E adesso so cosa devo fare, devo continuare a
> respirare perch?? domani il
> > sole sorger?? e chiss?? la marea cosa potr??
> portare. (Cast Away)
> > Il progresso! Sempre tardi arriva. (Nuovo Cinema
> Paradiso)
> >
> 
> 
> 
> -- 
> Ci?? che facciamo in vita riecheggia
> nell'eternit??...(Il Gladiatore)
> "Noodles, cos'hai fatto in tutti questi anni?" "
> Sono andato a letto
> presto." (C'era una volta in America)
> E adesso so cosa devo fare, devo continuare a
> respirare perch?? domani il
> sole sorger?? e chiss?? la marea cosa potr??
> portare. (Cast Away)
> Il progresso! Sempre tardi arriva. (Nuovo Cinema
> Paradiso)
> > _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options,
> please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
> 




 
____________________________________________________________________________________
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Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users.
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From sakcahalit at hotmail.com  Tue Mar 20 21:13:09 2007
From: sakcahalit at hotmail.com (Halit Sakca)
Date: Tue, 20 Mar 2007 23:13:09 +0200
Subject: [SIPForum-discussion] List of registered user on SIP server
Message-ID: <BAY126-W208EA6B32EB35C7A85835CC750@phx.gbl>

hi Pavan,
In my opinion,
1.) Best way to know how many users are online is querying the database,
2.) On the other side you can use specific debugging tools that can debug the stack(or SIP application server), as far as I know it can turn the number of registered users or call numbers.
 But of course I dont know that your application stack has this tool, program etc...
hopefully, answer was useful :)
 
Halit,


From: pavan.jain at ingenient.comTo: francesco.landolfo at gmail.comDate: Tue, 20 Mar 2007 10:03:49 -0400CC: discussion at sipforum.orgSubject: Re: [SIPForum-discussion] List of registered user on SIP server








Our product needs to know how many users are online (registered) just like we have in IM chat sessions. For that I was wondering my SIP stack can send a Query message from stack to Server, and get a list of registered users. 
 




From: Francesco Paolo Landolfo [mailto:francesco.landolfo at gmail.com] Sent: Tuesday, March 20, 2007 6:24 AMTo: Pavan JainCc: wellya; discussion at sipforum.orgSubject: Re: [SIPForum-discussion] List of registered user on SIP server
 
I think no. Why would you try this message? In what scenario would you use it?

2007/3/19, Pavan Jain <pavan.jain at ingenient.com>: 


Is there a message format that I need to send to Server( similar to REGISTER)? An example will be helpful.
 
Regards,
Pavan
 




From: wellya [mailto:wellya at wellya.net] Sent: Saturday, March 17, 2007 2:25 AMTo: Pavan Jain; discussion at sipforum.orgSubject: Re: [SIPForum-discussion] List of registered user on SIP server
 

you can query database

  




wellya 

2007-03-17 





???? Pavan Jain

????? 2007-03-16 22:20:43

???? discussion at sipforum.org 

??? 

??? [SIPForum-discussion] List of registered user on SIP server

  

Hello All:
 
How can I get the list of registered users on the SIP server?
 
Regards,
Pavan
_______________________________________________This is the SIP Forum discussion mailing listTO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussionPost to the list at discussion at sipforum.org
-- Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto presto." (C'era una volta in America) E adesso so cosa devo fare, devo continuare a respirare perch? domani il sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso) 
_________________________________________________________________
Live.com'u deneyin - h?zl? ve ki?iselle?tirilmi? giri? sayfan?zla istedi?iniz her ?ey tek bir yerde.
http://www.live.com/getstarted
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From rakesh_rcm at yahoo.com  Tue Mar 20 21:31:44 2007
From: rakesh_rcm at yahoo.com (rakesh menon)
Date: Tue, 20 Mar 2007 14:31:44 -0700 (PDT)
Subject: [SIPForum-discussion] Pingtel
Message-ID: <706330.97117.qm@web56612.mail.re3.yahoo.com>

Hi all,

Does anyone work on Pingtel Softphones/Servers?

caio
Rakesh


 
____________________________________________________________________________________
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Check out "Tonight's Picks" on Yahoo! TV.
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From jainp1979 at gmail.com  Wed Mar 21 09:34:09 2007
From: jainp1979 at gmail.com (pankaj jain)
Date: Wed, 21 Mar 2007 15:04:09 +0530
Subject: [SIPForum-discussion] doubt in Example-3.7 of RFC 3665
Message-ID: <d41814cf0703210234y3b4c2916o5c6ed48e21328158@mail.gmail.com>

Hi,
I was going through RFC 3665 -- Basic Call Flow Examples: 3.7 - Session with
re-INVITE (IP Address
Change)<http://www.tech-invite.com/SIP/3665/Ti-SIP%203665-3.7%20%28v1.1%29.pdf>
The CSeq header in 1st INVITE (Alice to Bob) is 1
The CSeq header in 2nd INVITE (Bob to Alice) is 14
and The CSeq header in BYE (Alice to Bob) is 2

My questions are:
Shouldn't CSeq be monotonically increasing in a call?
is CSeq similar to TCP seq number where both parties maintain their own
sequence numbers?

-- 
Thanks
Pankaj Jain
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From rjsparks at nostrum.com  Wed Mar 21 10:05:49 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Wed, 21 Mar 2007 11:05:49 +0100
Subject: [SIPForum-discussion] doubt in Example-3.7 of RFC 3665
In-Reply-To: <d41814cf0703210234y3b4c2916o5c6ed48e21328158@mail.gmail.com>
References: <d41814cf0703210234y3b4c2916o5c6ed48e21328158@mail.gmail.com>
Message-ID: <45C80385-434E-4B2C-A647-E75A3C8D01FC@nostrum.com>

The CSeq sequence is scoped to each endpoint (Alice keeps a  
monotonically increasing sequence and Bob keeps a _separate_  
monotonically increasing sequence in this dialog).

RjS

On Mar 21, 2007, at 10:34 AM, pankaj jain wrote:

> Hi,
> I was going through RFC 3665 -- Basic Call Flow Examples: 3.7 -  
> Session with re-INVITE (IP Address Change)
> The CSeq header in 1st INVITE (Alice to Bob) is 1
> The CSeq header in 2nd INVITE (Bob to Alice) is 14
> and The CSeq header in BYE (Alice to Bob) is 2
>
> My questions are:
> Shouldn't CSeq be monotonically increasing in a call?
> is CSeq similar to TCP seq number where both parties maintain their  
> own sequence numbers?
>
> -- 
> Thanks
> Pankaj Jain
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit http:// 
> sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org

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From indresh.singh at siemens.com  Wed Mar 21 13:20:26 2007
From: indresh.singh at siemens.com (Singh, Indresh (SNL US))
Date: Wed, 21 Mar 2007 06:20:26 -0700
Subject: [SIPForum-discussion] doubt in Example-3.7 of RFC 3665
In-Reply-To: <45C80385-434E-4B2C-A647-E75A3C8D01FC@nostrum.com>
Message-ID: <3D80B10873C01D47BEC71C8DE311CF111C7ED2DD@USNWK100MSX.ww017.siemens.net>

Rob is absolutely right. Both parties maintain there own sequence number
as described in the dialog section ( 12.1 ) of RFC-3261
 
A dialog contains certain pieces of state needed for further message

transmissions within the dialog. This state consists of the dialog

ID, a local sequence number (used to order requests from the UA to

its peer), a remote sequence number (used to order requests from its

peer to the UA), a local URI, a remote URI, remote target, a boolean

flag called "secure", and a route set, which is an ordered list of

URIs. The route set is the list of servers that need to be traversed

to send a request to the peer. A dialog can also be in the "early"

state, which occurs when it is created with a provisional response,

and then transition to the "confirmed" state when a 2xx final

response arrives. For other responses, or if no response arrives at

all on that dialog, the early dialog terminates.

Regards,
 
Indresh K Singh

________________________________

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Robert Sparks
Sent: Wednesday, March 21, 2007 6:06 AM
To: pankaj jain
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] doubt in Example-3.7 of RFC 3665


The CSeq sequence is scoped to each endpoint (Alice keeps a
monotonically increasing sequence and Bob keeps a _separate_
monotonically increasing sequence in this dialog). 

RjS

On Mar 21, 2007, at 10:34 AM, pankaj jain wrote:


	Hi,
	I was going through RFC 3665 -- Basic Call Flow Examples: 3.7 -
Session with re-INVITE (IP Address Change)
<http://www.tech-invite.com/SIP/3665/Ti-SIP%203665-3.7%20%28v1.1%29.pdf>

	The CSeq header in 1st INVITE (Alice to Bob) is 1
	The CSeq header in 2nd INVITE (Bob to Alice) is 14
	and The CSeq header in BYE (Alice to Bob) is 2 
	
	My questions are:
	Shouldn't CSeq be monotonically increasing in a call?
	is CSeq similar to TCP seq number where both parties maintain
their own sequence numbers?
	
	-- 
	Thanks
	Pankaj Jain 
	_______________________________________________
	This is the SIP Forum discussion mailing list
	TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
	Post to the list at discussion at sipforum.org


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From Shanmukharao.Makkapati at airtel.in  Wed Mar 21 14:52:05 2007
From: Shanmukharao.Makkapati at airtel.in (Shanmukharao.Makkapati at airtel.in)
Date: Wed, 21 Mar 2007 20:22:05 +0530
Subject: [SIPForum-discussion] IM Using SIP ( Please Help Me )
Message-ID: <OF4F4355CC.256236DF-ON652572A5.005122B9-652572A5.00513511@bharti.com>


Dear Friends,
                  I want to know about Design process in SIP Im using C 
and Linux. If we get Instant messenging project using SIP on C & Linux 
Platform. How can we divide it into modules and aim of each module ( input 
and output )..

Like SIP Parser module. Can you just give me a brief about these 
please.......... Where SDP comes in, Where RTP Comes in, TCP, UDP Use... 
and The design process..., How these modules will be combined to totally 
form a IM Product. please........................please..........


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From adam.harding2 at ntlworld.com  Wed Mar 21 17:08:42 2007
From: adam.harding2 at ntlworld.com (Adam Harding)
Date: Wed, 21 Mar 2007 17:08:42 +0000
Subject: [SIPForum-discussion] Confused with RTP analysis function in
	Ethereal, please help!
Message-ID: <20070321161151.IQNC17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>

Hi,

This is more of an ethereal/wireshark question but is related to SIP as I am trying to analyse SIP calls:

Please could someone help me as I am quite confused!

What does "Delta" mean in the RTP analysis and how is it calculated?

In the RTP graph analysis, what does the red line indicating "Difference" mean and how is it calculated? 

I thought the "difference" on the graph was giving the Delta results in graph format but the results on the graph are different and lower than the Delta values. 

I am doing my final year project on VOIP and looking into how different network conditions effect the call quality. 

What measurement in wireshark should I be looking at to show the effect that causes bad voice quality due to the variation in packet arrivals at the recieving end? 

Any help would be great as I am very confused!

Thanks.

Adam.


-----------------------------------------
Email sent from www.virginmedia.com/email
Virus-checked using McAfee(R) Software and scanned for spam



From jani_tech_forum at yahoo.com  Wed Mar 21 16:23:05 2007
From: jani_tech_forum at yahoo.com (Janakiraman N)
Date: Wed, 21 Mar 2007 09:23:05 -0700 (PDT)
Subject: [SIPForum-discussion] How to send request to one proxy to another
Message-ID: <300031.93195.qm@web62108.mail.re1.yahoo.com>

Hi ALL,

We are developing conference application using SIP Servlet API (JSR 116 standard).
 
SIP soft phone sending INVITE and we are forwarding that INVITE request to another proxy by using pushRoute() method from SIPServletRequest interface and proxyTo() method from Proxy interface.
 
But SIP soft phone while sending BYE request to close the dialog, remote tag ("To" tag) does not match with 200 OK response which received for INVITE.
 
I can not able to find where the exact problem is located.
 
Please give your valuable suggestion.

Regards,
Janakiraman N


 
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From sukerry at 126.com  Thu Mar 22 04:25:45 2007
From: sukerry at 126.com (sukerry)
Date: Thu, 22 Mar 2007 12:25:45 +0800
Subject: [SIPForum-discussion] how to compute cps(call per second)?
Message-ID: <46020555.01275C.01564@m5-141.126.com>

Young-Geun Park,

Please use sipp and ethreal

======== 2007-03-20 19:31:43 you wrote========

and what are there performance factors related to a sip servlet container that deploys the proxy app?

Park



From: Young-Geun Park [mailto:zeroroot at tmax.co.kr] 
Sent: Tuesday, March 20, 2007 8:22 PM
To: 'discussion at sipforum.org'
Subject: [SIPForum-discussion] how to compute cps(call per second)?

Hi, all

I want to know how to compute CPS(call per second) specially with a proxyApp as follows.

UAC          Proxy         UAS
   1(invite)
 ----------?
   2(100 trying)
<-----------
   3(invite)
----------?
4(180 Ringing)
<-----------
5(180 Ringing)
<-----------
6(200 OK)
<-----------
7(200 OK)
<-----------
8(ACK)
----------?
   9(ACK)
----------?
10(BYE)
<-----------
11(BYE)
<-----------
12(200 OK)
----------?
   13(200 OK)
----------?

Thanks in advance

Regards, 
Park

= = = = = = = = = = = = = = = = = = = = = = 

??????????????sukerry
??????????????sukerry at 126.com
???????????????2007-03-22
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From pallavim35 at gmail.com  Thu Mar 22 05:10:07 2007
From: pallavim35 at gmail.com (aditi g)
Date: Thu, 22 Mar 2007 10:40:07 +0530
Subject: [SIPForum-discussion] Confused with RTP analysis function in
	Ethereal, please help!
In-Reply-To: <20070321161151.IQNC17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>
References: <20070321161151.IQNC17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>
Message-ID: <63af059d0703212210x78968163l30fcc10e3edc3867@mail.gmail.com>

Hi
Jitter measurement in wireshark shows the effect that causes bad voice
quality due to the variation in packet arrivals at the recieving
end.Acceptable jitter values are upto 40 ms ,if jitter exceeds 40 ms then
poor voice quality can be apprended.

Also delta is time interval between 2 packets which is apprx 20 ms.
If you sort packets in "RTP Analysis: window by Seq no ,you will be able to
see packetization interval
of appx 20 ms which is Delta in this case.

Thanks
Pallavi




On 3/21/07, Adam Harding <adam.harding2 at ntlworld.com> wrote:
>
> Hi,
>
> This is more of an ethereal/wireshark question but is related to SIP as I
> am trying to analyse SIP calls:
>
> Please could someone help me as I am quite confused!
>
> What does "Delta" mean in the RTP analysis and how is it calculated?
>
> In the RTP graph analysis, what does the red line indicating "Difference"
> mean and how is it calculated?
>
> I thought the "difference" on the graph was giving the Delta results in
> graph format but the results on the graph are different and lower than the
> Delta values.
>
> I am doing my final year project on VOIP and looking into how different
> network conditions effect the call quality.
>
> What measurement in wireshark should I be looking at to show the effect
> that causes bad voice quality due to the variation in packet arrivals at the
> recieving end?
>
> Any help would be great as I am very confused!
>
> Thanks.
>
> Adam.
>
>
> -----------------------------------------
> Email sent from www.virginmedia.com/email
> Virus-checked using McAfee(R) Software and scanned for spam
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
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From jainp1979 at gmail.com  Thu Mar 22 05:15:28 2007
From: jainp1979 at gmail.com (pankaj jain)
Date: Thu, 22 Mar 2007 10:45:28 +0530
Subject: [SIPForum-discussion] doubt in Example-3.7 of RFC 3665
In-Reply-To: <3D80B10873C01D47BEC71C8DE311CF111C7ED2DD@USNWK100MSX.ww017.siemens.net>
References: <45C80385-434E-4B2C-A647-E75A3C8D01FC@nostrum.com>
	<3D80B10873C01D47BEC71C8DE311CF111C7ED2DD@USNWK100MSX.ww017.siemens.net>
Message-ID: <d41814cf0703212215h1a3e1582l1f0b0ace5b7efbb6@mail.gmail.com>

Thanks a lot friends
The doubt is cleared now.

On 3/21/07, Singh, Indresh (SNL US) <indresh.singh at siemens.com> wrote:
>
>  Rob is absolutely right. Both parties maintain there own sequence number
> as described in the dialog section ( 12.1 ) of RFC-3261
>
>
> A dialog contains certain pieces of state needed for further message
>
> transmissions within the dialog. This state consists of the dialog
>
> ID, *a local sequence number (used to order requests from the UA to*
>
> *its peer), a remote sequence number (used to order requests from its*
>
> *peer to the UA),* a local URI, a remote URI, remote target, a boolean
>
> flag called "secure", and a route set, which is an ordered list of
>
> URIs. The route set is the list of servers that need to be traversed
>
> to send a request to the peer. A dialog can also be in the "early"
>
> state, which occurs when it is created with a provisional response,
>
> and then transition to the "confirmed" state when a 2xx final
>
> response arrives. For other responses, or if no response arrives at
>
> all on that dialog, the early dialog terminates.
> Regards,
>
> Indresh K Singh
>
>  ------------------------------
> *From:* discussion-bounces at sipforum.org [mailto:
> discussion-bounces at sipforum.org] *On Behalf Of *Robert Sparks
> *Sent:* Wednesday, March 21, 2007 6:06 AM
> *To:* pankaj jain
> *Cc:* discussion at sipforum.org
> *Subject:* Re: [SIPForum-discussion] doubt in Example-3.7 of RFC 3665
>
> The CSeq sequence is scoped to each endpoint (Alice keeps a monotonically
> increasing sequence and Bob keeps a _separate_ monotonically increasing
> sequence in this dialog).
> RjS
>
>  On Mar 21, 2007, at 10:34 AM, pankaj jain wrote:
>
> Hi,
> I was going through RFC 3665 -- Basic Call Flow Examples: 3.7 - Session
> with re-INVITE (IP Address Change)<http://www.tech-invite.com/SIP/3665/Ti-SIP%203665-3.7%20%28v1.1%29.pdf>
> The CSeq header in 1st INVITE (Alice to Bob) is 1
> The CSeq header in 2nd INVITE (Bob to Alice) is 14
> and The CSeq header in BYE (Alice to Bob) is 2
>
> My questions are:
> Shouldn't CSeq be monotonically increasing in a call?
> is CSeq similar to TCP seq number where both parties maintain their own
> sequence numbers?
>
> --
> Thanks
> Pankaj Jain _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
>


-- 
Thanks
Pankaj Jain
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From adam.harding2 at ntlworld.com  Thu Mar 22 12:38:41 2007
From: adam.harding2 at ntlworld.com (Adam Harding)
Date: Thu, 22 Mar 2007 12:38:41 +0000
Subject: [SIPForum-discussion] Simplified algorithm to give some sort of
	E-model result
Message-ID: <20070322123841.XPYB17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>

Hi,

Does anyone know on some sort of basic algorithm that I can put some VOIP analysis results of Jitter, Delay, Packet Loss etc into and get some sort of an estimate of Voice quality, like the R-factor?, but more simple, just to give a rough figure.

I have not got sufficient data or resources to use the proper E-model/MOS/R-Facor, but if there is some sort of free equation I can plug some basic results into just to get some sort of rough figure for voice quality that would be great.

At the moment I am writing a report for my University project and am just giving my opinion on how different factors effect the audio quality.

Just for the benefit of my examiner really, so I have some sort of numerical figure to use to compare the my results, rather than just my opinion of the audio quality.


Thanks,

Adam Harding

-----------------------------------------
Email sent from www.virginmedia.com/email
Virus-checked using McAfee(R) Software and scanned for spam



From gargashish11 at gmail.com  Thu Mar 22 14:03:34 2007
From: gargashish11 at gmail.com (Ashish Garg)
Date: Thu, 22 Mar 2007 19:33:34 +0530
Subject: [SIPForum-discussion] Query on SIPp
Message-ID: <f11c9280703220703o3cb85e8cv61001cb1c31d2e39@mail.gmail.com>

Hi,
 Does anyone know while using RTP echo feature of *SIPp* why the tool
echo the RTP/UDP packets back to the sender coming on the port specfied +2?

*Statements as in Documentation of SIPp:*

*The "RTP echo" feature allows SIPp to listen to one or two local IP address
and port (specified using -mi and -mp command line parameters) for RTP
media. Everything that is received on this address/port is echoed back to
the sender. *

*RTP/UDP packets coming on this port + 2 are also echoed to their sender
(used for sound and video echo).*

** Thanks

Ashish Garg
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From varlei.knupp at siemens.com  Thu Mar 22 15:30:32 2007
From: varlei.knupp at siemens.com (Knupp, Varlei Fernandes)
Date: Thu, 22 Mar 2007 12:30:32 -0300
Subject: [SIPForum-discussion] doubt about "contact field"
Message-ID: <52D77941ACCDE243874A51D2E17489540207F1E7@SAO1015V.ww101.siemens.net>



Hi to all,

I have a doubt about "contact field".

The proxy send a "INVITE" message =INVITE sip:33133010 at 192.168.2.100:5060 SIP/2.0 to Gateway

The gateway send  a "session progress" message to proxy, but this message has contact field = Contact: <sip:1100 at 192.168.2.100>

The User disconnect the call.

The proxy send a "CANCEL" message to gateway = CANCEL sip:1100 at 192.168.2.100:5060 SIP/2.0 

So the gateway reject the message.

Who is correct?
- The Gateway because reject the call because the proxy changed the header in the message CANCEL
- The Proxy because when the gateway sent contact field = <sip:1100 at 192.168.2.100> the header CANCEL message now is  CANCEL sip:1100 at 192.168.2.100:5060 SIP/2.0.

thanks a lot


Varlei Fernandes Knupp

SIEMENS Communications

Customer Services Enterprise

RCC S?o Paulo - Technical Support

Tel.: +55 11 3817 2659

Fax: +55 11 3817 2613

varlei.knupp at siemens.com <blocked::mailto:varlei.knupp at siemens.com>




________________________________________
The information contained in this e-mail is for the exclusive use of the intended recipient(s) and may be confidential, proprietary, and/or legally privileged.  Inadvertent disclosure of this message does not constitute a waiver of any privilege.  If you receive this message in error, please do not directly or indirectly use, print, copy, forward, or disclose any part of this message.  Please also delete this e-mail and all copies and notify the sender. Thank you. For alternate languages please go to http://www.siemens.com.ar/disclaimer/
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From Shanmukharao.Makkapati at airtel.in  Thu Mar 22 17:37:44 2007
From: Shanmukharao.Makkapati at airtel.in (Shanmukharao.Makkapati at airtel.in)
Date: Thu, 22 Mar 2007 23:07:44 +0530
Subject: [SIPForum-discussion] IM Design
Message-ID: <OF0E64B733.DF99E92A-ON652572A6.00608D89-652572A6.00605E49@bharti.com>


Hi to all,

When we design IM using SIP 

The modules may be, 

1.       we may have to have parser module (to parse incmonibg/outgoing 
msges),
2.       core call engine (in case if you want VOIP telephony),
3.       authentication modules to check the validity of the accounts,
4.       network module for data transfer (UDP for IM) and
5.       presence module to provide status of the user (online/offline)..

How these modules will be structered in a sequence to design IM.

Can any one provide me the aritecture of the same in breif please....





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From indresh.singh at siemens.com  Thu Mar 22 20:16:24 2007
From: indresh.singh at siemens.com (Singh, Indresh (SNL US))
Date: Thu, 22 Mar 2007 13:16:24 -0700
Subject: [SIPForum-discussion] doubt about "contact field"
In-Reply-To: <52D77941ACCDE243874A51D2E17489540207F1E7@SAO1015V.ww101.siemens.net>
Message-ID: <3D80B10873C01D47BEC71C8DE311CF111CB117AB@USNWK100MSX.ww017.siemens.net>

CANCEL message has to be exactly same as INVITE request and also should be sent to the same host port as original INVITE. So in this case gateway correctly rejects the message.

________________________________

From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Knupp, Varlei Fernandes
Sent: Thursday, March 22, 2007 11:31 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] doubt about "contact field"


 
Hi to all,
 
I have a doubt about "contact field".
 
The proxy send a "INVITE" message =INVITE sip:33133010 at 192.168.2.100:5060 SIP/2.0 to Gateway
 
The gateway send  a "session progress" message to proxy, but this message has contact field = Contact: <sip:1100 at 192.168.2.100>
 
The User disconnect the call.
 
The proxy send a "CANCEL" message to gateway = CANCEL sip:1100 at 192.168.2.100:5060 SIP/2.0  
 
So the gateway reject the message.
 
Who is correct? 
- The Gateway because reject the call because the proxy changed the header in the message CANCEL
- The Proxy because when the gateway sent contact field = <sip:1100 at 192.168.2.100> the header CANCEL message now is  CANCEL sip:1100 at 192.168.2.100:5060 SIP/2.0.
 
thanks a lot
 
 
Varlei Fernandes Knupp 

SIEMENS Communications

Customer Services Enterprise

RCC S?o Paulo - Technical Support

Tel.: +55 11 3817 2659 

Fax: +55 11 3817 2613

varlei.knupp at siemens.com <blocked::mailto:varlei.knupp at siemens.com> 

 
________________________________________
The information contained in this e-mail is for the exclusive use of the intended recipient(s) and may be confidential, proprietary, and/or legally privileged. Inadvertent disclosure of this message does not constitute a waiver of any privilege. If you receive this message in error, please do not directly or indirectly use, print, copy, forward, or disclose any part of this message. Please also delete this e-mail and all copies and notify the sender. Thank you. For alternate languages please go to http://www.siemens.com.ar/disclaimer/
	
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From varlei.knupp at siemens.com  Thu Mar 22 20:19:29 2007
From: varlei.knupp at siemens.com (Knupp, Varlei Fernandes)
Date: Thu, 22 Mar 2007 17:19:29 -0300
Subject: [SIPForum-discussion] doubt about "contact field"
In-Reply-To: <3D80B10873C01D47BEC71C8DE311CF111CB117AB@USNWK100MSX.ww017.siemens.net>
References: <52D77941ACCDE243874A51D2E17489540207F1E7@SAO1015V.ww101.siemens.net>
	<3D80B10873C01D47BEC71C8DE311CF111CB117AB@USNWK100MSX.ww017.siemens.net>
Message-ID: <52D77941ACCDE243874A51D2E17489540207F54B@SAO1015V.ww101.siemens.net>


Thanks a lot for your help.:-)


Varlei Fernandes Knupp

SIEMENS Communications

Customer Services Enterprise

RCC S?o Paulo - Technical Support

Tel.: +55 11 3817 2659

Fax: +55 11 3817 2613

varlei.knupp at siemens.com <blocked::mailto:varlei.knupp at siemens.com>



________________________________

From: Singh, Indresh (SNL US)
Sent: quinta-feira, 22 de mar?o de 2007 17:16
To: Knupp, Varlei Fernandes; discussion at sipforum.org
Subject: RE: [SIPForum-discussion] doubt about "contact field"


CANCEL message has to be exactly same as INVITE request and also should be sent to the same host port as original INVITE. So in this case gateway correctly rejects the message.

________________________________

From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Knupp, Varlei Fernandes
Sent: Thursday, March 22, 2007 11:31 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] doubt about "contact field"



Hi to all,

I have a doubt about "contact field".

The proxy send a "INVITE" message =INVITE sip:33133010 at 192.168.2.100:5060 SIP/2.0 to Gateway

The gateway send  a "session progress" message to proxy, but this message has contact field = Contact: <sip:1100 at 192.168.2.100>

The User disconnect the call.

The proxy send a "CANCEL" message to gateway = CANCEL sip:1100 at 192.168.2.100:5060 SIP/2.0 

So the gateway reject the message.

Who is correct?
- The Gateway because reject the call because the proxy changed the header in the message CANCEL
- The Proxy because when the gateway sent contact field = <sip:1100 at 192.168.2.100> the header CANCEL message now is  CANCEL sip:1100 at 192.168.2.100:5060 SIP/2.0.

thanks a lot


Varlei Fernandes Knupp

SIEMENS Communications

Customer Services Enterprise

RCC S?o Paulo - Technical Support

Tel.: +55 11 3817 2659

Fax: +55 11 3817 2613

varlei.knupp at siemens.com <blocked::mailto:varlei.knupp at siemens.com>


________________________________________
The information contained in this e-mail is for the exclusive use of the intended recipient(s) and may be confidential, proprietary, and/or legally privileged. Inadvertent disclosure of this message does not constitute a waiver of any privilege. If you receive this message in error, please do not directly or indirectly use, print, copy, forward, or disclose any part of this message. Please also delete this e-mail and all copies and notify the sender. Thank you. For alternate languages please go to http://www.siemens.com.ar/disclaimer/



________________________________________
The information contained in this e-mail is for the exclusive use of the intended recipient(s) and may be confidential, proprietary, and/or legally privileged.  Inadvertent disclosure of this message does not constitute a waiver of any privilege.  If you receive this message in error, please do not directly or indirectly use, print, copy, forward, or disclose any part of this message.  Please also delete this e-mail and all copies and notify the sender. Thank you. For alternate languages please go to http://www.siemens.com.ar/disclaimer/
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From cmarzotta at gmail.com  Thu Mar 22 20:55:38 2007
From: cmarzotta at gmail.com (Claudio Marzotta)
Date: Thu, 22 Mar 2007 17:55:38 -0300
Subject: [SIPForum-discussion] contact field
In-Reply-To: <3D80B10873C01D47BEC71C8DE311CF111CB117AB@USNWK100MSX.ww017.siemens.net>
References: <52D77941ACCDE243874A51D2E17489540207F1E7@SAO1015V.ww101.siemens.net>
	<3D80B10873C01D47BEC71C8DE311CF111CB117AB@USNWK100MSX.ww017.siemens.net>
Message-ID: <005501c76cc4$73ce05a0$080000c8@bcdros.com>

 

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From danishzaidi54 at yahoo.com  Thu Mar 22 23:50:10 2007
From: danishzaidi54 at yahoo.com (Danish Zaidi)
Date: Thu, 22 Mar 2007 16:50:10 -0700 (PDT)
Subject: [SIPForum-discussion] am i missing something with call hold
Message-ID: <290796.4343.qm@web90604.mail.mud.yahoo.com>

Hello 

the Hold Event is sent like the INVITE the difference is only attribute is sendonly

if m correct then why its not working, this sendHold Code works if i use it for Invite Purpose only...

public void sendHold()
    {        
        try
        {
            toUser = "1112";
            outgoingCall = true;
            rtpConnection = new RtpConnection();
            localRtpPort = rtpConnection.inizialize(localRtpStartPort, localRtpEndPort, bufferLenght, minimumThreshold, enabledThreshold, packetSize);
            System.out.print("Sending INVITE...  ");
            reqUnauthInvite = null;
            reqAuthInvite = null;
            SipURI requestURI = addressFactory.createSipURI(toUser, serverIpPort);
            SipURI toAddress = addressFactory.createSipURI(toUser, serverIp);
            System.out.println("Server IP In Client.java is "+serverIp);
            Address toNameAddress = addressFactory.createAddress(toAddress);
            ToHeader toHeader = headerFactory.createToHeader(toNameAddress, null);
            System.out.println("IPAddress In Client.java is "+sipStack.getIPAddress());
            SipURI fromAddress = addressFactory.createSipURI(username, sipStack.getIPAddress());
            Address fromNameAddress = addressFactory.createAddress(fromAddress);
            FromHeader fromHeader = headerFactory.createFromHeader(fromNameAddress, userTag);
            ArrayList viaHeaders = new ArrayList();
            javax.sip.header.ViaHeader viaHeader = headerFactory.createViaHeader(sipStack.getIPAddress(), sipProvider.getListeningPoint().getPort(), transportProt, null);
            viaHeaders.add(viaHeader);
            CallIdHeader callIdHeader = cldTemp;
            CSeqHeader cSeqHeader = headerFactory.createCSeqHeader(1, "INVITE");
            MaxForwardsHeader maxForwards = headerFactory.createMaxForwardsHeader(70);
            Request request = messageFactory.createRequest(requestURI, "INVITE", callIdHeader, cSeqHeader, fromHeader, toHeader, viaHeaders, maxForwards);
            request.addHeader(contactHeader);
            ContentTypeHeader contentTypeHeader = headerFactory.createContentTypeHeader("application", "sdp");
            String myAddress = Globals.addr.getHostAddress();            
            String string1 = " RTP/AVP";
            String string2 = "";
            int i=0;
            //for(int i = 0; i < codecListModel.getSize(); i++)
            {
                if(String.valueOf(codecListModel.elementAt(i)).equals(" PCMU/8000 "))
                {
                    string1 = (new StringBuilder()).append(string1).append(" 0").toString();
                    string2 = (new StringBuilder()).append(string2).append("a=rtpmap:0 PCMU/8000\r\n").toString();
                    //continue;
                }
                if(String.valueOf(codecListModel.elementAt(i)).equals(" GSM/8000 "))
                {
                    string1 = (new StringBuilder()).append(string1).append(" 3").toString();
                    string2 = (new StringBuilder()).append(string2).append("a=rtpmap:3 GSM/8000\r\n").toString();
                    //continue;
                }
                if(String.valueOf(codecListModel.elementAt(i)).equals(" G723/8000 "))
                {
                    string1 = (new StringBuilder()).append(string1).append(" 4").toString();
                    string2 = (new StringBuilder()).append(string2).append("a=rtpmap:4 G723/8000\r\n").toString();
                    //continue;
                }
                if(String.valueOf(codecListModel.elementAt(i)).equals(" DVI4/8000 "))
                {
                    string1 = (new StringBuilder()).append(string1).append(" 5").toString();
                    string2 = (new StringBuilder()).append(string2).append("a=rtpmap:5 DVI4/8000\r\n").toString();
                }
            }

            string2+="a=sendonly";
            string1 = (new StringBuilder()).append(string1).append("\r\n").toString();
            String sdpData = (new StringBuilder()).append("v=0\r\no=4855 13760799956958020 13760799956958020 IN IP4  ").append(myAddress).append("\r\n").append("s=Session SDP\r\n").append("c=IN IP4  ").append(myAddress).append("\r\n").append("t=0 0\r\n").append("m=audio ").append(localRtpPort).append(string1).append(string2).toString();
            byte contents[] = sdpData.getBytes();
            request.setContent(contents, contentTypeHeader);
            request.addHeader(userAgentHeader);
            javax.sip.header.Header callInfoHeader = headerFactory.createHeader("Call-Info", "<http://www.kleis.it>");
            request.addHeader(callInfoHeader);
            inviteTid = sipProvider.getNewClientTransaction(request);
            inviteTid.sendRequest();
            dialog = inviteTid.getDialog();
            reqUnauthInvite = request;
            System.out.println("DONE");
        }
        catch(Exception ex)
        {
            System.out.println(ex.getMessage());
            ex.printStackTrace();
        }
        return;
    }


sorrry about the indentation mistakes

but plzz help me with the SIP Call HOLD

thanx in advance

 
---------------------------------
Bored stiff? Loosen up...
Download and play hundreds of games for free on Yahoo! Games.
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From peter324.kim at samsung.com  Fri Mar 23 00:27:45 2007
From: peter324.kim at samsung.com (HYUNGON KIM)
Date: Fri, 23 Mar 2007 00:27:45 +0000 (GMT)
Subject: [SIPForum-discussion] Unscribe
Message-ID: <0JFB00IP2YM9VE@ms5.samsung.com>

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From peter324.kim at samsung.com  Fri Mar 23 00:29:46 2007
From: peter324.kim at samsung.com (HYUNGON KIM)
Date: Fri, 23 Mar 2007 00:29:46 +0000 (GMT)
Subject: [SIPForum-discussion] Unsubscribe
Message-ID: <0JFB00MH8YPMQW@ms5.samsung.com>

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From ramank24 at gmail.com  Fri Mar 23 04:13:05 2007
From: ramank24 at gmail.com (raman kumar)
Date: Fri, 23 Mar 2007 04:13:05 +0000
Subject: [SIPForum-discussion] how to compute cps(call per second)?
In-Reply-To: <46020555.01275C.01564@m5-141.126.com>
References: <46020555.01275C.01564@m5-141.126.com>
Message-ID: <67002eb30703222113h242732d6n613e36cea0331719@mail.gmail.com>

caps defines the load capacity of sip proxy.
It is number of INVITE message it can handle without with reply of
TRYING 100 message without droping any call at same time( call set up
time may be different )

eg if your SIP proxy is capable of responding to 100 INVITE message
per second ( think a phone is sending 100 INVITE per second and is
capable of making 100 connections )

These things can also be tested using sipp tool

On 22/03/07, sukerry <sukerry at 126.com> wrote:
> Young-Geun Park,
>
> Please use sipp and ethreal
>
> ======== 2007-03-20 19:31:43 you wrote========
>
> and what are there performance factors related to a sip servlet container
> that deploys the proxy app?
>
> Park
>
>
>
> From: Young-Geun Park [mailto:zeroroot at tmax.co.kr]
> Sent: Tuesday, March 20, 2007 8:22 PM
> To: 'discussion at sipforum.org'
> Subject: [SIPForum-discussion] how to compute cps(call per second)?
>
> Hi, all
>
> I want to know how to compute CPS(call per second) specially with a proxyApp
> as follows.
>
> UAC          Proxy         UAS
>    1(invite)
>  ----------?
>    2(100 trying)
> <-----------
>    3(invite)
> ----------?
> 4(180 Ringing)
> <-----------
> 5(180 Ringing)
> <-----------
> 6(200 OK)
> <-----------
> 7(200 OK)
> <-----------
> 8(ACK)
> ----------?
>    9(ACK)
> ----------?
> 10(BYE)
> <-----------
> 11(BYE)
> <-----------
> 12(200 OK)
> ----------?
>    13(200 OK)
> ----------?
>
> Thanks in advance
>
> Regards,
> Park
>
> = = = = = = = = = = = = = = = = = = = = = =
>
> ??????????????sukerry
> ??????????????sukerry at 126.com
> ???????????????2007-03-22
>



From skp10_9559 at yahoo.co.in  Fri Mar 23 04:32:13 2007
From: skp10_9559 at yahoo.co.in (santosh patra)
Date: Fri, 23 Mar 2007 10:02:13 +0530 (IST)
Subject: [SIPForum-discussion] Add in Contact List
Message-ID: <653482.24520.qm@web8812.mail.in.yahoo.com>




		
__________________________________________________________
Yahoo! India Answers: Share what you know. Learn something new
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From vtsoukanov at jnetx.ru  Fri Mar 23 07:53:25 2007
From: vtsoukanov at jnetx.ru (Victor Tsoukanov)
Date: Fri, 23 Mar 2007 10:53:25 +0300
Subject: [SIPForum-discussion] IM Design
References: <OF0E64B733.DF99E92A-ON652572A6.00608D89-652572A6.00605E49@bharti.com>
Message-ID: <00ba01c76d20$57276890$d800a8c0@jnetx.ru>


----- Original Message ----- 
From: Shanmukharao.Makkapati at airtel.in
To: discussion at sipforum.org
Sent: Thursday, March 22, 2007 8:37 PM
Subject: [SIPForum-discussion] IM Design



Hi to all,

When we design IM using SIP

The modules may be,

1.         we may have to have parser module (to parse incmonibg/outgoing
msges),
2.         core call engine (in case if you want VOIP telephony),
3.         authentication modules to check the validity of the accounts,
4.         network module for data transfer (UDP for IM) and
5.         presence module to provide status of the user (online/offline)..

How these modules will be structered in a sequence to design IM.

Can any one provide me the aritecture of the same in breif please....

Hi

Look at SIP servlet API. First and fourth items already implemetnted in SIP
servlet container, other ones can be implemented with different servlets.
For example, second item (as I understand it is call control) - can be
implemeted with B2B servlet, authentication module and presence module - is
a simple proxy servlets. All this stuff can be collected in one application
with simple dispatcher servlet or controller servlet (there are a lot of
suitable examples for HTTP servlets).



From rjsparks at nostrum.com  Fri Mar 23 08:08:06 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Fri, 23 Mar 2007 09:08:06 +0100
Subject: [SIPForum-discussion] SIPit 20 registration closes March 30
Message-ID: <1CC1DF4D-00A0-4DE8-8E63-610B80C488BC@nostrum.com>

SIPit 20 registration closes one week from today (March 30).

The event will be in Antwerp, Belgium hosted by Alcatel-Lucent.

If you are planning to attend, but have not registered, please do so  
now.
Information and a link to the registration site is available at
http://www.sipit.net

RjS


From pallavim35 at gmail.com  Fri Mar 23 08:22:10 2007
From: pallavim35 at gmail.com (aditi g)
Date: Fri, 23 Mar 2007 13:52:10 +0530
Subject: [SIPForum-discussion] Unsubscribe
In-Reply-To: <0JFB00MH8YPMQW@ms5.samsung.com>
References: <0JFB00MH8YPMQW@ms5.samsung.com>
Message-ID: <63af059d0703230122w5f696d01u528b3d3d3d82a840@mail.gmail.com>

Hello

Here when you are sending Invite to hold the call,it is Reinvite that you
have to send.
So from code ,i could see that you are creating new client
transaction.Thisis wrong as for hold ,
you have send REinvite with same sip header as orginal invite but with
different SDP where you can
specify a= sendonly.You do not have to create new transaction.
Thanks


On 3/23/07, HYUNGON KIM <peter324.kim at samsung.com> wrote:
>
> Unsubscribe
>
>
>
>
>
>
> ------- *Original Message* -------
> *Sender* : Danish Zaidi<danishzaidi54 at yahoo.com>
> *Date* : 2007-03-23 08:50
> *Title* : [SIPForum-discussion] am i missing something with call hold
>
> Hello
>
> the Hold Event is sent like the INVITE the difference is only attribute is
> sendonly
>
> if m correct then why its not working, this sendHold Code works if i use
> it for Invite Purpose only...
>
> public void sendHold()
>     {
>         try
>         {
>             toUser = "1112";
>             outgoingCall = true;
>             rtpConnection = new RtpConnection();
>             localRtpPort = rtpConnection.inizialize(localRtpStartPort,
> localRtpEndPort, bufferLenght, minimumThreshold, enabledThreshold,
> packetSize);
>             System.out.print("Sending INVITE...  ");
>             reqUnauthInvite = null;
>             reqAuthInvite = null;
>             SipURI requestURI = addressFactory.createSipURI(toUser,
> serverIpPort);
>             SipURI toAddress = addressFactory.createSipURI(toUser,
> serverIp);
>             System.out.println("Server IP In Client.java is "+serverIp);
>             Address toNameAddress = addressFactory.createAddress
> (toAddress);
>             ToHeader toHeader = headerFactory.createToHeader(toNameAddress,
> null);
>             System.out.println("IPAddress In Client.java is
> "+sipStack.getIPAddress());
>             SipURI fromAddress = addressFactory.createSipURI(username,
> sipStack.getIPAddress());
>             Address fromNameAddress = addressFactory.createAddress
> (fromAddress);
>             FromHeader fromHeader = headerFactory.createFromHeader(fromNameAddress,
> userTag);
>             ArrayList viaHeaders = new ArrayList();
>             javax.sip.header.ViaHeader viaHeader =
> headerFactory.createViaHeader(sipStack.getIPAddress(),
> sipProvider.getListeningPoint().getPort(), transportProt, null);
>             viaHeaders.add(viaHeader);
>             CallIdHeader callIdHeader = cldTemp;
>             CSeqHeader cSeqHeader = headerFactory.createCSeqHeader(1,
> "INVITE");
>             MaxForwardsHeader maxForwards =
> headerFactory.createMaxForwardsHeader(70);
>             Request request = messageFactory.createRequest(requestURI,
> "INVITE", callIdHeader, cSeqHeader, fromHeader, toHeader, viaHeaders,
> maxForwards);
>             request.addHeader(contactHeader);
>             ContentTypeHeader contentTypeHeader =
> headerFactory.createContentTypeHeader("application", "sdp");
>             String myAddress = Globals.addr.getHostAddress();
>             String string1 = " RTP/AVP";
>             String string2 = "";
>             int i=0;
>             //for(int i = 0; i < codecListModel.getSize(); i++)
>             {
>                 if(String.valueOf(codecListModel.elementAt(i)).equals("
> PCMU/8000 "))
>                 {
>                     string1 = (new
> StringBuilder()).append(string1).append(" 0").toString();
>                     string2 = (new
> StringBuilder()).append(string2).append("a=rtpmap:0
> PCMU/8000\r\n").toString();
>                     //continue;
>                 }
>                 if(String.valueOf(codecListModel.elementAt(i)).equals("
> GSM/8000 "))
>                 {
>                     string1 = (new
> StringBuilder()).append(string1).append(" 3").toString();
>                     string2 = (new
> StringBuilder()).append(string2).append("a=rtpmap:3
> GSM/8000\r\n").toString();
>                     //continue;
>                 }
>                 if(String.valueOf(codecListModel.elementAt(i)).equals("
> G723/8000 "))
>                 {
>                     string1 = (new
> StringBuilder()).append(string1).append(" 4").toString();
>                     string2 = (new
> StringBuilder()).append(string2).append("a=rtpmap:4
> G723/8000\r\n").toString();
>                     //continue;
>                 }
>                 if(String.valueOf(codecListModel.elementAt(i)).equals("
> DVI4/8000 "))
>                 {
>                     string1 = (new
> StringBuilder()).append(string1).append(" 5").toString();
>                     string2 = (new
> StringBuilder()).append(string2).append("a=rtpmap:5
> DVI4/8000\r\n").toString();
>                 }
>             }
>
>             string2+="a=sendonly";
>             string1 = (new
> StringBuilder()).append(string1).append("\r\n").toString();
>             String sdpData = (new StringBuilder()).append("v=0\r\no=4855
> 13760799956958020 13760799956958020 IN IP4
> ").append(myAddress).append("\r\n").append("s=Session SDP\r\n").append("c=IN
> IP4  ").append(myAddress).append("\r\n").append("t=0 0\r\n").append("m=audio
> ").append(localRtpPort).append(string1).append(string2).toString();
>             byte contents[] = sdpData.getBytes();
>             request.setContent(contents, contentTypeHeader);
>             request.addHeader(userAgentHeader);
>             javax.sip.header.Header callInfoHeader =
> headerFactory.createHeader("Call-Info", "<http://www.kleis.it>");
>             request.addHeader(callInfoHeader);
>             inviteTid = sipProvider.getNewClientTransaction(request);
>             inviteTid.sendRequest();
>             dialog = inviteTid.getDialog();
>             reqUnauthInvite = request;
>             System.out.println("DONE");
>         }
>         catch(Exception ex)
>         {
>             System.out.println(ex.getMessage());
>             ex.printStackTrace();
>         }
>         return;
>     }
>
>
> sorrry about the indentation mistakes
>
> but plzz help me with the SIP Call HOLD
>
> thanx in advance
>
> ------------------------------
> Bored stiff? <http://us.rd.yahoo.com/evt=49935/*http://games.yahoo.com>Loosen up...
> Download and play hundreds of games for free<http://us.rd.yahoo.com/evt=49935/*http://games.yahoo.com>on Yahoo! Games.
>
>
>
>
>        <http://www.samsungnetworks.co.kr/>   ??? (KIM HYUN GON) ??
> ?????? ??3? Tel. 070-7015-0442 Fax.070-7015-5678 M.P  017-642-1518
> peter324.kim at samsung.com    <http://www.wyz070.co.kr/>
>
>
>
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
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From yong2.chen at siemens.com  Fri Mar 23 08:22:24 2007
From: yong2.chen at siemens.com (Chen, Yong SNLB PEK)
Date: Fri, 23 Mar 2007 16:22:24 +0800
Subject: [SIPForum-discussion] Unscribe
Message-ID: <FD353C9F0D65454D911AFD6D3669ADC405693633@PEKW934A.cn001.siemens.net>

Unscribe
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From Alois.Komenda at R-KOM.de  Fri Mar 23 09:42:53 2007
From: Alois.Komenda at R-KOM.de (Komenda Alois)
Date: Fri, 23 Mar 2007 10:42:53 +0100
Subject: [SIPForum-discussion] Experience with SIP Application Servers
Message-ID: <2A5386A266D8DB11A4AE0090275130BC06A17D@ffserver>

Hello,

has anyone of you experience on working with one of these Application
Servers: ApexVoice OmniVox3D, Aricent SIP AS, BEA WebLogic SIP, Pactolus
RapidFLEX, Ubiquity SIP AS?

Can you tell me about strength and weaknesses of these Servers, supported
features and developing new services?

Best regards

Alois Komenda



From Rakesh.Hooda at aricent.com  Fri Mar 23 10:09:27 2007
From: Rakesh.Hooda at aricent.com (Rakesh Hooda)
Date: Fri, 23 Mar 2007 15:39:27 +0530
Subject: [SIPForum-discussion] Experience with SIP Application Servers
Message-ID: <OFEB66B940.6DC47CF1-ON652572A7.0037CC31-652572A7.0037CC35@flextronicssoftware.com>

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From francesco.landolfo at gmail.com  Fri Mar 23 10:59:07 2007
From: francesco.landolfo at gmail.com (Francesco Paolo Landolfo)
Date: Fri, 23 Mar 2007 11:59:07 +0100
Subject: [SIPForum-discussion] How to implement a chat M2M
Message-ID: <d08f948c0703230359h55727018x62fe914d05c96de1@mail.gmail.com>

Hi,
I have implemented a chat P2P using SIP and the MESSAGE message.
Pratically, if Pippo want to send an instant message to Pluto, Pippo sends a
MESSAGE to a Server, this Server convert the Pluto Sip Uri in Pluto Care of
Address and send it to Pluto. This is in agreement with rfc3261.

Now I'd like to implement a chat M2M using SIP, but I have some doubt about
flow and messages that I have to use.

I think that there is one possible scene:

   - A client send a MESSAGE to a Server that dispatchs this message to
   all chat partecipant.

Now, how can I implement this feature?
Have someone some other idea?

Thanks,
Francesco

-- 
Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
"Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto
presto." (C'era una volta in America)
E adesso so cosa devo fare, devo continuare a respirare perch? domani il
sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso)
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From qt.kiran at gmail.com  Fri Mar 23 11:19:10 2007
From: qt.kiran at gmail.com (kiran chakkilam)
Date: Fri, 23 Mar 2007 16:49:10 +0530
Subject: [SIPForum-discussion] pls help me in this basic scenario
Message-ID: <e0646ce70703230419j7910efb4r389f27d192cea550@mail.gmail.com>

Hi everybody,


I am currently working on the sipp1.1 . i want to test proxy server(System
Under test). so i am simulating UAC and UAS using sipp. i will

send invite messge from UAC ,it will hit proxy(SUT).Proxy responds with 100
trying response back to UAC with all headers in short format

then my UAC is terminated.It's showing an error  No valid call-id. so I have
some doubts in sipp.

1. whether sipp accepts headers in shorcode format.

2. why it's terminating after receiving the 100 trying response.


so kindly help me in this scenario,it's very urgent for me


Thanks & Regards

ch.kiran
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From pallavim35 at gmail.com  Fri Mar 23 08:33:22 2007
From: pallavim35 at gmail.com (aditi g)
Date: Fri, 23 Mar 2007 14:03:22 +0530
Subject: [SIPForum-discussion] Simplified algorithm to give some sort of
	E-model result
In-Reply-To: <20070322123841.XPYB17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>
References: <20070322123841.XPYB17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>
Message-ID: <63af059d0703230133t3dd262aaibe8f50302a16bc14@mail.gmail.com>

Hello,
I am sending document as attachment that shows calcualtion of traffic
metrics.
Thanks
Pallavi


On 3/22/07, Adam Harding <adam.harding2 at ntlworld.com> wrote:
>
> Hi,
>
> Does anyone know on some sort of basic algorithm that I can put some VOIP
> analysis results of Jitter, Delay, Packet Loss etc into and get some sort of
> an estimate of Voice quality, like the R-factor?, but more simple, just to
> give a rough figure.
>
> I have not got sufficient data or resources to use the proper
> E-model/MOS/R-Facor, but if there is some sort of free equation I can plug
> some basic results into just to get some sort of rough figure for voice
> quality that would be great.
>
> At the moment I am writing a report for my University project and am just
> giving my opinion on how different factors effect the audio quality.
>
> Just for the benefit of my examiner really, so I have some sort of
> numerical figure to use to compare the my results, rather than just my
> opinion of the audio quality.
>
>
> Thanks,
>
> Adam Harding
>
> -----------------------------------------
> Email sent from www.virginmedia.com/email
> Virus-checked using McAfee(R) Software and scanned for spam
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
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From chris_christophersen at hotmail.com  Sat Mar 24 18:15:37 2007
From: chris_christophersen at hotmail.com (Chris Christophersen)
Date: Sat, 24 Mar 2007 14:15:37 -0400
Subject: [SIPForum-discussion] (no subject)
Message-ID: <BAY103-F14F32D527D6CD6A8736A5EFC690@phx.gbl>

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From deepanshu at huawei.com  Mon Mar 26 01:39:25 2007
From: deepanshu at huawei.com (Deepanshu)
Date: Mon, 26 Mar 2007 09:39:25 +0800
Subject: [SIPForum-discussion] How to implement a chat M2M
References: <d08f948c0703230359h55727018x62fe914d05c96de1@mail.gmail.com>
Message-ID: <00c301c76f47$96a7f7e0$8178a40a@china.huawei.com>

You can use the concept defined in draft-ietf-sipping-capacity-attribute-03.txt.

Or 

A Adhoc conference can be established by Client A and all other users can join in the conference. They can exchange the message using MSRP



HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC


  ----- Original Message ----- 
  From: Francesco Paolo Landolfo 
  To: discussion at sipforum.org 
  Sent: Friday, March 23, 2007 6:59 PM
  Subject: [SIPForum-discussion] How to implement a chat M2M


  Hi,
  I have implemented a chat P2P using SIP and the MESSAGE message. 
  Pratically, if Pippo want to send an instant message to Pluto, Pippo sends a MESSAGE to a Server, this Server convert the Pluto Sip Uri in Pluto Care of Address and send it to Pluto. This is in agreement with rfc3261. 

  Now I'd like to implement a chat M2M using SIP, but I have some doubt about flow and messages that I have to use.

  I think that there is one possible scene:

    a.. A client send a MESSAGE to a Server that dispatchs this message to all chat partecipant. 
  Now, how can I implement this feature? 
  Have someone some other idea?

  Thanks,
  Francesco

  -- 
  Ci? che facciamo in vita riecheggia nell'eternit?...(Il Gladiatore)
  "Noodles, cos'hai fatto in tutti questi anni?" " Sono andato a letto presto." (C'era una volta in America) 
  E adesso so cosa devo fare, devo continuare a respirare perch? domani il sole sorger? e chiss? la marea cosa potr? portare. (Cast Away)
  Il progresso! Sempre tardi arriva. (Nuovo Cinema Paradiso) 


------------------------------------------------------------------------------


  _______________________________________________
  This is the SIP Forum discussion mailing list
  TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
  Post to the list at discussion at sipforum.org
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From Shanmukharao.Makkapati at airtel.in  Mon Mar 26 08:29:13 2007
From: Shanmukharao.Makkapati at airtel.in (Shanmukharao.Makkapati at airtel.in)
Date: Mon, 26 Mar 2007 13:59:13 +0530
Subject: [SIPForum-discussion] Interview Faqs
Message-ID: <OF05C5D884.B386EE92-ON652572AA.002E871B-652572AA.002E3F76@bharti.com>


Dear All,
           Can anyone send me the interview faqs on sip protocol...Please


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From umair3210 at yahoo.com  Mon Mar 26 11:18:06 2007
From: umair3210 at yahoo.com (Muhammad Umair)
Date: Mon, 26 Mar 2007 04:18:06 -0700 (PDT)
Subject: [SIPForum-discussion] softphone's help needed
Message-ID: <362122.81927.qm@web38714.mail.mud.yahoo.com>

hi all,
   
  i m umair , i m a student of final year computer science n IT. i m going to make my final project on soft phones. can any one tell me where to start from, practical implementations of SIP n some thing about practical developement environment of softphones.
   
  thank u

 
---------------------------------
8:00? 8:25? 8:40?  Find a flick in no time
 with theYahoo! Search movie showtime shortcut.
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From niklas.fondberg at tilgin.com  Mon Mar 26 19:30:24 2007
From: niklas.fondberg at tilgin.com (Niklas Fondberg)
Date: Mon, 26 Mar 2007 21:30:24 +0200
Subject: [SIPForum-discussion] simultaneous INVITEs
Message-ID: <1174937424.5519.12.camel@localhost.localdomain>

Hi, 
I new to this list but I hope that the list is what I'm after; an
implementation and design discussion list about SIP.
If my question is wrongly addressed, please forgive me and please point
me the right direction...

My question that I have searched all over for an answer to is quite
simple:

What is the correct behavior for a UA if a second INVITE arrives before
the first has been answered?

SIP Flow:

-------------> INVITE (1)
<------------- 100 Trying (1)
<------------- 180 Ringing (1)
-------------> INVITE (2)
... ???

Here the first (1) INVITE could have been answered by some other UA that
the INVITE might have been forked to and in this case session (2) should
start ringing. 


Niklas Fondberg





From adam.harding2 at ntlworld.com  Mon Mar 26 22:01:13 2007
From: adam.harding2 at ntlworld.com (Adam Harding)
Date: Mon, 26 Mar 2007 23:01:13 +0100
Subject: [SIPForum-discussion] R-Factor type equation to evaluate VOIP
	quality from Wireshark RTP stats
Message-ID: <20070326220113.LRUQ17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>

Hi,

I am interested in any free algorithms that can be used to give a value for the voice quality in a VOIP call based on basic statistics such as delay, packet loss and jitter.

I asked this question a few days and got a really useful document recommended to me which helps me understand how the R-Factor works but I can not get hold of the ITU-G values and my RTP results from Wireshark are probably to basic to calculate the R-Factor.

So just wondering if there is some sort of basic algorithm that I can enter my results from the Wireshark RTP stats and get some sort of value of voice quality that I can use to compare my results with each other. 

Thanks,

Adam Harding

-----------------------------------------
Email sent from www.virginmedia.com/email
Virus-checked using McAfee(R) Software and scanned for spam



From deepanshu at huawei.com  Tue Mar 27 01:50:17 2007
From: deepanshu at huawei.com (Deepanshu)
Date: Tue, 27 Mar 2007 09:50:17 +0800
Subject: [SIPForum-discussion] simultaneous INVITEs
References: <1174937424.5519.12.camel@localhost.localdomain>
Message-ID: <003801c77012$45a10050$8178a40a@china.huawei.com>

inline
----- Original Message -----
From: "Niklas Fondberg" <niklas.fondberg at tilgin.com>
To: <discussion at sipforum.org>
Sent: Tuesday, March 27, 2007 3:30 AM
Subject: [SIPForum-discussion] simultaneous INVITEs


> Hi,
> I new to this list but I hope that the list is what I'm after; an
> implementation and design discussion list about SIP.
> If my question is wrongly addressed, please forgive me and please point
> me the right direction...
>
> My question that I have searched all over for an answer to is quite
> simple:
>
> What is the correct behavior for a UA if a second INVITE arrives before
> the first has been answered?
>
> SIP Flow:
>
> -------------> INVITE (1)
> <------------- 100 Trying (1)
> <------------- 180 Ringing (1)
> -------------> INVITE (2)
> ... ???
>
> Here the first (1) INVITE could have been answered by some other UA that
> the INVITE might have been forked to and in this case session (2) should
> start ringing.

If the first (1) INVITE could have been answered by some other UA then the
originating UAC SHALL send a CANCEL request towards UAS instead of a INVITE
(2).

-------------> INVITE (1)
<------------- 100 Trying (1)
<------------- 180 Ringing (1)
--------------> CANCEL (1)
                        stop ringing
<------------- SIP 487 (1)


HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC

>
>
> Niklas Fondberg
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>





From belic_sonja at yahoo.com  Tue Mar 27 15:05:23 2007
From: belic_sonja at yahoo.com (Sonja Belic)
Date: Tue, 27 Mar 2007 08:05:23 -0700 (PDT)
Subject: [SIPForum-discussion] Authentication and authorization in SIP
Message-ID: <664094.34155.qm@web60623.mail.yahoo.com>

Hi,
 I have a question regarding authentication and authorization mechanism in SIP. For instance, if there are more then one applications running on the same SIP system, does every application authenticate itself or all of them use the same authentication parameters, defined for that SIP system?
 Thanks in advance.
 
 Best Regards,
 Sonja
 
---------------------------------
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with Yahoo! Mail for Mobile. Get started.
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From qt.kiran at gmail.com  Tue Mar 27 16:25:47 2007
From: qt.kiran at gmail.com (kiran chakkilam)
Date: Tue, 27 Mar 2007 21:55:47 +0530
Subject: [SIPForum-discussion] content length
Message-ID: <e0646ce70703270925i183a21faocb49e8158212f9b2@mail.gmail.com>

Hi all,


How to calculate message body in sipp. Any open source tools are there to
calculate message body. please help me . It's very urgent for me.






Thanks in Advance

ch.kiran
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From adam.harding2 at ntlworld.com  Tue Mar 27 19:02:25 2007
From: adam.harding2 at ntlworld.com (Adam Harding)
Date: Tue, 27 Mar 2007 20:02:25 +0100
Subject: [SIPForum-discussion] RTP Delta and Difference : whats the
	difference?!
Message-ID: <20070327190225.BWMA17393.aamtaout02-winn.ispmail.ntl.com@smtp.ntlworld.com>

Hi,

Regarding RTP analysis in Wireshark/Ethereal I am confused between the difference between "Delta" which I think is the delay between 2 consecutive packets and the "difference" value that is indicated on the graph given in ethereal. 

How is the difference value indicated on the graph calculated? Is there a forumla?

Thanks.

-----------------------------------------
Email sent from www.virginmedia.com/email
Virus-checked using McAfee(R) Software and scanned for spam



From wang.ran at byd.com.cn  Wed Mar 28 03:31:43 2007
From: wang.ran at byd.com.cn (wangran)
Date: Wed, 28 Mar 2007 11:31:43 +0800
Subject: [SIPForum-discussion]
	=?gb2312?B?tPC4tDogZGlzY3Vzc2lvbiBEaWdlc3QsIFZvbCAyMCwgSXNzdWUgMzg=?=
In-Reply-To: <mailman.1.1175011201.26118.discussion@sipforum.org>
Message-ID: <OF0ECD4452.9BFB2601-ON482572AC.00137279@byd.com.cn>

Hi..
         We have some question in RFC3261, May I have you some minute to
explain it?
         Alice and bob?s call flaw in chapter 4 figure.1 is like this:

         SIP Flow:

-------------> INVITE (F1)\

-------------> INVITE (F2)

<------------- 100 Trying (F3)

         

         But in chapter 24.2

         F2 and F3 exchange there sequence

-------------> INVITE (F1)\

<------------- 100 Trying (F2)

-------------> INVITE (F3)

 

Does this small difference cause problems?

your comment will be highly appreciated!

 


Best of Regards,
 
wangran
 
 ***********************************************************************
BYD TECHFAITH?COMPANY?LIMITED(BTC)
 
Address:3/F,M8 West,NO.1 Jiu Xian Qiao Dong Road,
        Chao Yang District,Beijing,China  100016
PostCode:10016
Mobile: +86-13810362150
Tel: +86-10-58291226
Mail: wang.ran at byd.com.cn
 ***********************************************************************
 
 
-----????-----
???: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.
org] ?? discussion-request at sipforum.org
????: 2007?3?28? 0:00
???: discussion at sipforum.org
??: discussion Digest, Vol 20, Issue 38

Send discussion mailing list submissions to
	discussion at sipforum.org

To subscribe or unsubscribe via the World Wide Web, visit
	http://sipforum.org/mailman/listinfo/discussion
or, via email, send a message with subject or body 'help' to
	discussion-request at sipforum.org

You can reach the person managing the list at
	discussion-owner at sipforum.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of discussion digest..."


Today's Topics:

   1. simultaneous INVITEs (Niklas Fondberg)
   2. R-Factor type equation to evaluate VOIP	quality from
      Wireshark RTP stats (Adam Harding)
   3. Re: simultaneous INVITEs (Deepanshu)
   4. Authentication and authorization in SIP (Sonja Belic)


----------------------------------------------------------------------

Message: 1
Date: Mon, 26 Mar 2007 21:30:24 +0200
From: Niklas Fondberg <niklas.fondberg at tilgin.com>
Subject: [SIPForum-discussion] simultaneous INVITEs
To: discussion at sipforum.org
Message-ID: <1174937424.5519.12.camel at localhost.localdomain>
Content-Type: text/plain

Hi, 
I new to this list but I hope that the list is what I'm after; an
implementation and design discussion list about SIP.
If my question is wrongly addressed, please forgive me and please point
me the right direction...

My question that I have searched all over for an answer to is quite
simple:

What is the correct behavior for a UA if a second INVITE arrives before
the first has been answered?

SIP Flow:

-------------> INVITE (1)
<------------- 100 Trying (1)
<------------- 180 Ringing (1)
-------------> INVITE (2)
... ???

Here the first (1) INVITE could have been answered by some other UA that
the INVITE might have been forked to and in this case session (2) should
start ringing. 


Niklas Fondberg





------------------------------

Message: 2
Date: Mon, 26 Mar 2007 23:01:13 +0100
From: Adam Harding <adam.harding2 at ntlworld.com>
Subject: [SIPForum-discussion] R-Factor type equation to evaluate VOIP
	quality from Wireshark RTP stats
To: "discussion at sipforum.org" <discussion at sipforum.org>
Message-ID:
	
<20070326220113.LRUQ17393.aamtaout02-winn.ispmail.ntl.com at smtp.ntlworld.com>
	
Content-Type: text/plain; charset=ISO-8859-1

Hi,

I am interested in any free algorithms that can be used to give a value for
the voice quality in a VOIP call based on basic statistics such as delay,
packet loss and jitter.

I asked this question a few days and got a really useful document
recommended to me which helps me understand how the R-Factor works but I can
not get hold of the ITU-G values and my RTP results from Wireshark are
probably to basic to calculate the R-Factor.

So just wondering if there is some sort of basic algorithm that I can enter
my results from the Wireshark RTP stats and get some sort of value of voice
quality that I can use to compare my results with each other. 

Thanks,

Adam Harding

-----------------------------------------
Email sent from www.virginmedia.com/email
Virus-checked using McAfee(R) Software and scanned for spam



------------------------------

Message: 3
Date: Tue, 27 Mar 2007 09:50:17 +0800
From: Deepanshu <deepanshu at huawei.com>
Subject: Re: [SIPForum-discussion] simultaneous INVITEs
To: Niklas Fondberg <niklas.fondberg at tilgin.com>
Cc: discussion at sipforum.org
Message-ID: <003801c77012$45a10050$8178a40a at china.huawei.com>
Content-Type: text/plain; charset=iso-8859-1

inline
----- Original Message -----
From: "Niklas Fondberg" <niklas.fondberg at tilgin.com>
To: <discussion at sipforum.org>
Sent: Tuesday, March 27, 2007 3:30 AM
Subject: [SIPForum-discussion] simultaneous INVITEs


> Hi,
> I new to this list but I hope that the list is what I'm after; an
> implementation and design discussion list about SIP.
> If my question is wrongly addressed, please forgive me and please point
> me the right direction...
>
> My question that I have searched all over for an answer to is quite
> simple:
>
> What is the correct behavior for a UA if a second INVITE arrives before
> the first has been answered?
>
> SIP Flow:
>
> -------------> INVITE (1)
> <------------- 100 Trying (1)
> <------------- 180 Ringing (1)
> -------------> INVITE (2)
> ... ???
>
> Here the first (1) INVITE could have been answered by some other UA that
> the INVITE might have been forked to and in this case session (2) should
> start ringing.

If the first (1) INVITE could have been answered by some other UA then the
originating UAC SHALL send a CANCEL request towards UAS instead of a INVITE
(2).

-------------> INVITE (1)
<------------- 100 Trying (1)
<------------- 180 Ringing (1)
--------------> CANCEL (1)
                        stop ringing
<------------- SIP 487 (1)


HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC

>
>
> Niklas Fondberg
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>





------------------------------

Message: 4
Date: Tue, 27 Mar 2007 08:05:23 -0700 (PDT)
From: Sonja Belic <belic_sonja at yahoo.com>
Subject: [SIPForum-discussion] Authentication and authorization in SIP
To: discussion at sipforum.org
Message-ID: <664094.34155.qm at web60623.mail.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi,
 I have a question regarding authentication and authorization mechanism in
SIP. For instance, if there are more then one applications running on the
same SIP system, does every application authenticate itself or all of them
use the same authentication parameters, defined for that SIP system?
 Thanks in advance.
 
 Best Regards,
 Sonja
 
---------------------------------
No need to miss a message. Get email on-the-go 
with Yahoo! Mail for Mobile. Get started.
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TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.
org/mailman/listinfo/discussion
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******************************************


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From pallavim35 at gmail.com  Wed Mar 28 04:22:26 2007
From: pallavim35 at gmail.com (aditi g)
Date: Wed, 28 Mar 2007 09:52:26 +0530
Subject: [SIPForum-discussion] Why Ack is different transaction?
Message-ID: <63af059d0703272122l5134afpa51426b1551dd36@mail.gmail.com>

Hi,

I want to know why ACK is considered different transaction from Invite
transaction .

regs
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From skp10_9559 at yahoo.co.in  Wed Mar 28 04:29:13 2007
From: skp10_9559 at yahoo.co.in (santosh patra)
Date: Wed, 28 Mar 2007 09:59:13 +0530 (IST)
Subject: [SIPForum-discussion] pls help me in this basic scenario
Message-ID: <318897.10226.qm@web8809.mail.in.yahoo.com>

Hello Kiran

As far as my knowledge says SIPP does not support any header except SDP headers in shortcode format,it can be if you are able to change the source code of SIPp and might be calleg is hanging that's why your call is not getting terminated.

Regards
Santosh Patra


----- Original Message ----
From: kiran chakkilam <qt.kiran at gmail.com>
To: discussion at sipforum.org
Sent: Friday, 23 March, 2007 4:49:10 PM
Subject: [SIPForum-discussion] pls help me in this basic scenario


Hi everybody,
 
 
I am currently working on the sipp1.1 . i want to test proxy server(System Under test). so i am simulating UAC and UAS using sipp. i will 
 
send invite messge from UAC ,it will hit proxy(SUT).Proxy responds with 100 trying response back to UAC with all headers in short format 
 
then my UAC is terminated.It's showing an error  No valid call-id. so I have some doubts in sipp.
 
1. whether sipp accepts headers in shorcode format.
 
2. why it's terminating after receiving the 100 trying response.
 
 
so kindly help me in this scenario,it's very urgent for me
 
 
Thanks & Regards
 
ch.kiran
 
 
_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org


		
__________________________________________________________
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http://in.answers.yahoo.com/
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From skp10_9559 at yahoo.co.in  Wed Mar 28 04:35:51 2007
From: skp10_9559 at yahoo.co.in (santosh patra)
Date: Wed, 28 Mar 2007 10:05:51 +0530 (IST)
Subject: [SIPForum-discussion] simultaneous INVITEs
Message-ID: <657666.6270.qm@web8807.mail.in.yahoo.com>

Hello Niklas

In this scenario, lets take a pratical example that
1. A calls B
2. B is ringing
3. In the mean time if C calls B then though A and B's call has not established, C should get busy response or if the A and B's call is established and B has call waiting activated then C han hear the Ringing of B user and B will get the indication that somebody is  calliing.

Santosh


----- Original Message ----
From: Niklas Fondberg <niklas.fondberg at tilgin.com>
To: discussion at sipforum.org
Sent: Tuesday, 27 March, 2007 1:00:24 AM
Subject: [SIPForum-discussion] simultaneous INVITEs


Hi, 
I new to this list but I hope that the list is what I'm after; an
implementation and design discussion list about SIP.
If my question is wrongly addressed, please forgive me and please point
me the right direction...

My question that I have searched all over for an answer to is quite
simple:

What is the correct behavior for a UA if a second INVITE arrives before
the first has been answered?

SIP Flow:

-------------> INVITE (1)
<------------- 100 Trying (1)
<------------- 180 Ringing (1)
-------------> INVITE (2)
... ???

Here the first (1) INVITE could have been answered by some other UA that
the INVITE might have been forked to and in this case session (2) should
start ringing. 


Niklas Fondberg



_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org


		
__________________________________________________________
Yahoo! India Answers: Share what you know. Learn something new
http://in.answers.yahoo.com/
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From deepanshu at huawei.com  Wed Mar 28 05:58:41 2007
From: deepanshu at huawei.com (Deepanshu)
Date: Wed, 28 Mar 2007 13:58:41 +0800
Subject: [SIPForum-discussion]
	=?gb2312?B?UmU6IFtTSVBGb3J1bS1kaXNjdXNzaW9uXbTwuLQ6IGRpc2N1c3Npbw==?=	=?gb2312?B?biBEaWdlc3QsIFZvbCAyMCwgSXNzdWUgMzg=?=
References: <OF0ECD4452.9BFB2601-ON482572AC.00137279@byd.com.cn>
Message-ID: <00b601c770fe$23466ad0$8178a40a@china.huawei.com>

Dear Wang

I don't think this create any kind of problems. 100 trying is hop-by-hop,
the proxy can perform it simultaneously with other request (INVITE P1--->P2
in your case)

HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC

----- Original Message -----
From: "wangran" <wang.ran at byd.com.cn>
To: <discussion at sipforum.org>
Sent: Wednesday, March 28, 2007 11:31 AM
Subject: [SIPForum-discussion]??: discussion Digest, Vol 20, Issue 38


> Hi..
>          We have some question in RFC3261, May I have you some minute to
> explain it?
>          Alice and bob?s call flaw in chapter 4 figure.1 is like this:
>
>          SIP Flow:
>
> -------------> INVITE (F1)\
>
> -------------> INVITE (F2)
>
> <------------- 100 Trying (F3)
>
>
>
>          But in chapter 24.2
>
>          F2 and F3 exchange there sequence
>
> -------------> INVITE (F1)\
>
> <------------- 100 Trying (F2)
>
> -------------> INVITE (F3)
>
>
>
> Does this small difference cause problems?
>
> your comment will be highly appreciated!
>
>
>
>
> Best of Regards,
>
> wangran
>
>  ***********************************************************************
> BYD TECHFAITH?COMPANY?LIMITED(BTC)
>
> Address:3/F,M8 West,NO.1 Jiu Xian Qiao Dong Road,
>         Chao Yang District,Beijing,China  100016
> PostCode:10016
> Mobile: +86-13810362150
> Tel: +86-10-58291226
> Mail: wang.ran at byd.com.cn
>  ***********************************************************************
>
>
> -----????-----
> ???: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.
> org] ?? discussion-request at sipforum.org
> ????: 2007?3?28? 0:00
> ???: discussion at sipforum.org
> ??: discussion Digest, Vol 20, Issue 38
>
> Send discussion mailing list submissions to
> discussion at sipforum.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://sipforum.org/mailman/listinfo/discussion
> or, via email, send a message with subject or body 'help' to
> discussion-request at sipforum.org
>
> You can reach the person managing the list at
> discussion-owner at sipforum.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of discussion digest..."
>
>
> Today's Topics:
>
>    1. simultaneous INVITEs (Niklas Fondberg)
>    2. R-Factor type equation to evaluate VOIP quality from
>       Wireshark RTP stats (Adam Harding)
>    3. Re: simultaneous INVITEs (Deepanshu)
>    4. Authentication and authorization in SIP (Sonja Belic)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 26 Mar 2007 21:30:24 +0200
> From: Niklas Fondberg <niklas.fondberg at tilgin.com>
> Subject: [SIPForum-discussion] simultaneous INVITEs
> To: discussion at sipforum.org
> Message-ID: <1174937424.5519.12.camel at localhost.localdomain>
> Content-Type: text/plain
>
> Hi,
> I new to this list but I hope that the list is what I'm after; an
> implementation and design discussion list about SIP.
> If my question is wrongly addressed, please forgive me and please point
> me the right direction...
>
> My question that I have searched all over for an answer to is quite
> simple:
>
> What is the correct behavior for a UA if a second INVITE arrives before
> the first has been answered?
>
> SIP Flow:
>
> -------------> INVITE (1)
> <------------- 100 Trying (1)
> <------------- 180 Ringing (1)
> -------------> INVITE (2)
> ... ???
>
> Here the first (1) INVITE could have been answered by some other UA that
> the INVITE might have been forked to and in this case session (2) should
> start ringing.
>
>
> Niklas Fondberg
>
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Mon, 26 Mar 2007 23:01:13 +0100
> From: Adam Harding <adam.harding2 at ntlworld.com>
> Subject: [SIPForum-discussion] R-Factor type equation to evaluate VOIP
> quality from Wireshark RTP stats
> To: "discussion at sipforum.org" <discussion at sipforum.org>
> Message-ID:
>
>
<20070326220113.LRUQ17393.aamtaout02-winn.ispmail.ntl.com at smtp.ntlworld.com>
>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hi,
>
> I am interested in any free algorithms that can be used to give a value
for
> the voice quality in a VOIP call based on basic statistics such as delay,
> packet loss and jitter.
>
> I asked this question a few days and got a really useful document
> recommended to me which helps me understand how the R-Factor works but I
can
> not get hold of the ITU-G values and my RTP results from Wireshark are
> probably to basic to calculate the R-Factor.
>
> So just wondering if there is some sort of basic algorithm that I can
enter
> my results from the Wireshark RTP stats and get some sort of value of
voice
> quality that I can use to compare my results with each other.
>
> Thanks,
>
> Adam Harding
>
> -----------------------------------------
> Email sent from www.virginmedia.com/email
> Virus-checked using McAfee(R) Software and scanned for spam
>
>
>
> ------------------------------
>
> Message: 3
> Date: Tue, 27 Mar 2007 09:50:17 +0800
> From: Deepanshu <deepanshu at huawei.com>
> Subject: Re: [SIPForum-discussion] simultaneous INVITEs
> To: Niklas Fondberg <niklas.fondberg at tilgin.com>
> Cc: discussion at sipforum.org
> Message-ID: <003801c77012$45a10050$8178a40a at china.huawei.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> inline
> ----- Original Message -----
> From: "Niklas Fondberg" <niklas.fondberg at tilgin.com>
> To: <discussion at sipforum.org>
> Sent: Tuesday, March 27, 2007 3:30 AM
> Subject: [SIPForum-discussion] simultaneous INVITEs
>
>
> > Hi,
> > I new to this list but I hope that the list is what I'm after; an
> > implementation and design discussion list about SIP.
> > If my question is wrongly addressed, please forgive me and please point
> > me the right direction...
> >
> > My question that I have searched all over for an answer to is quite
> > simple:
> >
> > What is the correct behavior for a UA if a second INVITE arrives before
> > the first has been answered?
> >
> > SIP Flow:
> >
> > -------------> INVITE (1)
> > <------------- 100 Trying (1)
> > <------------- 180 Ringing (1)
> > -------------> INVITE (2)
> > ... ???
> >
> > Here the first (1) INVITE could have been answered by some other UA that
> > the INVITE might have been forked to and in this case session (2) should
> > start ringing.
>
> If the first (1) INVITE could have been answered by some other UA then the
> originating UAC SHALL send a CANCEL request towards UAS instead of a
INVITE
> (2).
>
> -------------> INVITE (1)
> <------------- 100 Trying (1)
> <------------- 180 Ringing (1)
> --------------> CANCEL (1)
>                         stop ringing
> <------------- SIP 487 (1)
>
>
> HTH
>
> Deepanshu Gautam
> R&D Engineer
> Huawei Technologies Co. Ltd.
> Nanjing, PRC
>
> >
> >
> > Niklas Fondberg
> >
> >
> >
> > _______________________________________________
> > This is the SIP Forum discussion mailing list
> > TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> > Post to the list at discussion at sipforum.org
> >
>
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 27 Mar 2007 08:05:23 -0700 (PDT)
> From: Sonja Belic <belic_sonja at yahoo.com>
> Subject: [SIPForum-discussion] Authentication and authorization in SIP
> To: discussion at sipforum.org
> Message-ID: <664094.34155.qm at web60623.mail.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>  I have a question regarding authentication and authorization mechanism in
> SIP. For instance, if there are more then one applications running on the
> same SIP system, does every application authenticate itself or all of them
> use the same authentication parameters, defined for that SIP system?
>  Thanks in advance.
>
>  Best Regards,
>  Sonja
>
> ---------------------------------
> No need to miss a message. Get email on-the-go
> with Yahoo! Mail for Mobile. Get started.
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> URL:
>
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> hment-0001.html
>
> ------------------------------
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.
> org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
> End of discussion Digest, Vol 20, Issue 38
> ******************************************
>
>
> Powered by BYD Security Gateway.
>
>
>
> Powered by BYD Security Gateway.
>
>
>


----------------------------------------------------------------------------
----


> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
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>




From resalath.ahamed at gmail.com  Wed Mar 28 06:23:21 2007
From: resalath.ahamed at gmail.com (resalath ahamed)
Date: Wed, 28 Mar 2007 11:53:21 +0530
Subject: [SIPForum-discussion]
	=?GB2312?B?UmU6IFtTSVBGb3J1bS1kaXNjdXNzaW9uXSBSZTogW1NJUEZvcnVtLWRpc2N1?=
	=?GB2312?B?c3Npb25dtPC4tDogZGlzY3Vzc2lvbiBEaWdlc3QsIFZvbCAyMCwgSXNzdWUgMzg=?=
In-Reply-To: <00b601c770fe$23466ad0$8178a40a@china.huawei.com>
References: <OF0ECD4452.9BFB2601-ON482572AC.00137279@byd.com.cn>
	<00b601c770fe$23466ad0$8178a40a@china.huawei.com>
Message-ID: <3ec935670703272323pd28519cr9efcfe802c277034@mail.gmail.com>

wangran,

Your question can be solved by understanding "STATEFULL PROXY" and
"STATELESS PROXY".

Below are two scenarios:

[1] Call involves stateless proxy - Invite will be forwarded by a stateless
proxy without returning 100 trying to the originator. So in this case 100
trying should come from terminator or from other statefull proxy in the
network. The stateless proxy will forward the 100 trying to UAC. A stateless
proxy does not maintain the call state so it can not send 100 trying by its
own. It will only forward the 100 trying.

[2] Call involves statefull proxy - In this case a statefull proxy will
return 100 trying to the UAC before it forwards the request to the next SIP
element in the network. A statefull proxy maintains the call state and has
all the records of the call that is being processed. So it can trigger 100
trying to the originator.

Hope this solves your issue.

Thanks and Regards,
Resalath Ahamed.



On 3/28/07, Deepanshu <deepanshu at huawei.com> wrote:
>
> Dear Wang
>
> I don't think this create any kind of problems. 100 trying is hop-by-hop,
> the proxy can perform it simultaneously with other request (INVITE
> P1--->P2
> in your case)
>
> HTH
>
> Deepanshu Gautam
> R&D Engineer
> Huawei Technologies Co. Ltd.
> Nanjing, PRC
>
> ----- Original Message -----
> From: "wangran" <wang.ran at byd.com.cn>
> To: <discussion at sipforum.org>
> Sent: Wednesday, March 28, 2007 11:31 AM
> Subject: [SIPForum-discussion]??: discussion Digest, Vol 20, Issue 38
>
>
> > Hi..
> >          We have some question in RFC3261, May I have you some minute to
> > explain it?
> >          Alice and bob's call flaw in chapter 4 figure.1 is like this:
> >
> >          SIP Flow:
> >
> > -------------> INVITE (F1)\
> >
> > -------------> INVITE (F2)
> >
> > <------------- 100 Trying (F3)
> >
> >
> >
> >          But in chapter 24.2
> >
> >          F2 and F3 exchange there sequence
> >
> > -------------> INVITE (F1)\
> >
> > <------------- 100 Trying (F2)
> >
> > -------------> INVITE (F3)
> >
> >
> >
> > Does this small difference cause problems?
> >
> > your comment will be highly appreciated!
> >
> >
> >
> >
> > Best of Regards,
> >
> > wangran
> >
> >  ***********************************************************************
> > BYD TECHFAITH COMPANY LIMITED(BTC)
> >
> > Address:3/F,M8 West,NO.1 Jiu Xian Qiao Dong Road,
> >         Chao Yang District,Beijing,China  100016
> > PostCode:10016
> > Mobile: +86-13810362150
> > Tel: +86-10-58291226
> > Mail: wang.ran at byd.com.cn
> >  ***********************************************************************
> >
> >
> > -----????-----
> > ???: discussion-bounces at sipforum.org
> [mailto:discussion-bounces at sipforum.
> > org] ?? discussion-request at sipforum.org
> > ????: 2007?3?28? 0:00
> > ???: discussion at sipforum.org
> > ??: discussion Digest, Vol 20, Issue 38
> >
> > Send discussion mailing list submissions to
> > discussion at sipforum.org
> >
> > To subscribe or unsubscribe via the World Wide Web, visit
> > http://sipforum.org/mailman/listinfo/discussion
> > or, via email, send a message with subject or body 'help' to
> > discussion-request at sipforum.org
> >
> > You can reach the person managing the list at
> > discussion-owner at sipforum.org
> >
> > When replying, please edit your Subject line so it is more specific
> > than "Re: Contents of discussion digest..."
> >
> >
> > Today's Topics:
> >
> >    1. simultaneous INVITEs (Niklas Fondberg)
> >    2. R-Factor type equation to evaluate VOIP quality from
> >       Wireshark RTP stats (Adam Harding)
> >    3. Re: simultaneous INVITEs (Deepanshu)
> >    4. Authentication and authorization in SIP (Sonja Belic)
> >
> >
> > ----------------------------------------------------------------------
> >
> > Message: 1
> > Date: Mon, 26 Mar 2007 21:30:24 +0200
> > From: Niklas Fondberg <niklas.fondberg at tilgin.com>
> > Subject: [SIPForum-discussion] simultaneous INVITEs
> > To: discussion at sipforum.org
> > Message-ID: <1174937424.5519.12.camel at localhost.localdomain>
> > Content-Type: text/plain
> >
> > Hi,
> > I new to this list but I hope that the list is what I'm after; an
> > implementation and design discussion list about SIP.
> > If my question is wrongly addressed, please forgive me and please point
> > me the right direction...
> >
> > My question that I have searched all over for an answer to is quite
> > simple:
> >
> > What is the correct behavior for a UA if a second INVITE arrives before
> > the first has been answered?
> >
> > SIP Flow:
> >
> > -------------> INVITE (1)
> > <------------- 100 Trying (1)
> > <------------- 180 Ringing (1)
> > -------------> INVITE (2)
> > ... ???
> >
> > Here the first (1) INVITE could have been answered by some other UA that
> > the INVITE might have been forked to and in this case session (2) should
> > start ringing.
> >
> >
> > Niklas Fondberg
> >
> >
> >
> >
> >
> > ------------------------------
> >
> > Message: 2
> > Date: Mon, 26 Mar 2007 23:01:13 +0100
> > From: Adam Harding <adam.harding2 at ntlworld.com>
> > Subject: [SIPForum-discussion] R-Factor type equation to evaluate VOIP
> > quality from Wireshark RTP stats
> > To: "discussion at sipforum.org" <discussion at sipforum.org>
> > Message-ID:
> >
> >
> <
> 20070326220113.LRUQ17393.aamtaout02-winn.ispmail.ntl.com at smtp.ntlworld.com
> >
> >
> > Content-Type: text/plain; charset=ISO-8859-1
> >
> > Hi,
> >
> > I am interested in any free algorithms that can be used to give a value
> for
> > the voice quality in a VOIP call based on basic statistics such as
> delay,
> > packet loss and jitter.
> >
> > I asked this question a few days and got a really useful document
> > recommended to me which helps me understand how the R-Factor works but I
> can
> > not get hold of the ITU-G values and my RTP results from Wireshark are
> > probably to basic to calculate the R-Factor.
> >
> > So just wondering if there is some sort of basic algorithm that I can
> enter
> > my results from the Wireshark RTP stats and get some sort of value of
> voice
> > quality that I can use to compare my results with each other.
> >
> > Thanks,
> >
> > Adam Harding
> >
> > -----------------------------------------
> > Email sent from www.virginmedia.com/email
> > Virus-checked using McAfee(R) Software and scanned for spam
> >
> >
> >
> > ------------------------------
> >
> > Message: 3
> > Date: Tue, 27 Mar 2007 09:50:17 +0800
> > From: Deepanshu <deepanshu at huawei.com>
> > Subject: Re: [SIPForum-discussion] simultaneous INVITEs
> > To: Niklas Fondberg <niklas.fondberg at tilgin.com>
> > Cc: discussion at sipforum.org
> > Message-ID: <003801c77012$45a10050$8178a40a at china.huawei.com>
> > Content-Type: text/plain; charset=iso-8859-1
> >
> > inline
> > ----- Original Message -----
> > From: "Niklas Fondberg" <niklas.fondberg at tilgin.com>
> > To: <discussion at sipforum.org>
> > Sent: Tuesday, March 27, 2007 3:30 AM
> > Subject: [SIPForum-discussion] simultaneous INVITEs
> >
> >
> > > Hi,
> > > I new to this list but I hope that the list is what I'm after; an
> > > implementation and design discussion list about SIP.
> > > If my question is wrongly addressed, please forgive me and please
> point
> > > me the right direction...
> > >
> > > My question that I have searched all over for an answer to is quite
> > > simple:
> > >
> > > What is the correct behavior for a UA if a second INVITE arrives
> before
> > > the first has been answered?
> > >
> > > SIP Flow:
> > >
> > > -------------> INVITE (1)
> > > <------------- 100 Trying (1)
> > > <------------- 180 Ringing (1)
> > > -------------> INVITE (2)
> > > ... ???
> > >
> > > Here the first (1) INVITE could have been answered by some other UA
> that
> > > the INVITE might have been forked to and in this case session (2)
> should
> > > start ringing.
> >
> > If the first (1) INVITE could have been answered by some other UA then
> the
> > originating UAC SHALL send a CANCEL request towards UAS instead of a
> INVITE
> > (2).
> >
> > -------------> INVITE (1)
> > <------------- 100 Trying (1)
> > <------------- 180 Ringing (1)
> > --------------> CANCEL (1)
> >                         stop ringing
> > <------------- SIP 487 (1)
> >
> >
> > HTH
> >
> > Deepanshu Gautam
> > R&D Engineer
> > Huawei Technologies Co. Ltd.
> > Nanjing, PRC
> >
> > >
> > >
> > > Niklas Fondberg
> > >
> > >
> > >
> > > _______________________________________________
> > > This is the SIP Forum discussion mailing list
> > > TO UNSUBSCRIBE, or edit your delivery options, please visit
> > http://sipforum.org/mailman/listinfo/discussion
> > > Post to the list at discussion at sipforum.org
> > >
> >
> >
> >
> >
> >
> > ------------------------------
> >
> > Message: 4
> > Date: Tue, 27 Mar 2007 08:05:23 -0700 (PDT)
> > From: Sonja Belic <belic_sonja at yahoo.com>
> > Subject: [SIPForum-discussion] Authentication and authorization in SIP
> > To: discussion at sipforum.org
> > Message-ID: <664094.34155.qm at web60623.mail.yahoo.com>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Hi,
> >  I have a question regarding authentication and authorization mechanism
> in
> > SIP. For instance, if there are more then one applications running on
> the
> > same SIP system, does every application authenticate itself or all of
> them
> > use the same authentication parameters, defined for that SIP system?
> >  Thanks in advance.
> >
> >  Best Regards,
> >  Sonja
> >
> > ---------------------------------
> > No need to miss a message. Get email on-the-go
> > with Yahoo! Mail for Mobile. Get started.
> > -------------- next part --------------
> > An HTML attachment was scrubbed...
> > URL:
> >
>
> http://sipforum.org/pipermail/discussion/attachments/20070327/0fca17b7/attac
> > hment-0001.html
> >
> > ------------------------------
> >
> > _______________________________________________
> > This is the SIP Forum discussion mailing list
> > TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.
> > org/mailman/listinfo/discussion
> > Post to the list at discussion at sipforum.org
> >
> >
> > End of discussion Digest, Vol 20, Issue 38
> > ******************************************
> >
> >
> > Powered by BYD Security Gateway.
> >
> >
> >
> > Powered by BYD Security Gateway.
> >
> >
> >
>
>
>
> ----------------------------------------------------------------------------
> ----
>
>
> > _______________________________________________
> > This is the SIP Forum discussion mailing list
> > TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> > Post to the list at discussion at sipforum.org
> >
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
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>
>
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From avishekchowdhury at tataelxsi.co.in  Wed Mar 28 06:37:35 2007
From: avishekchowdhury at tataelxsi.co.in (Avishek Chowdhury)
Date: Wed, 28 Mar 2007 12:07:35 +0530
Subject: [SIPForum-discussion] Why Ack is different transaction?
References: <63af059d0703272122l5134afpa51426b1551dd36@mail.gmail.com>
Message-ID: <0fd201c77103$92b192a0$6c19320a@telxsi.com>

Hi Aditi,

When the UAC receives 200 OK, the client transaction is destroyed at TL.
This is done because, the TL is unaware of the above core. The above core can be a Proxy or an UAC core.
In case of proxy, the 200 OK is forwarded whereas in case of UAC, an ACK is generated. Now this ACK has to create a new transaction (transaction created by INVITE had been destroyed)
at TL for its transmission, hence the ACK for 200 OK is outside the INVITE transaction.

For other non-2xx final responses, the client transaction at TL is not destroyed and the ACK is generated by TL. Hence in this case, the ACK is within the transaction.

Regards,
Avishek

  ----- Original Message ----- 
  From: aditi g 
  To: discussion at sipforum.org 
  Sent: Wednesday, March 28, 2007 9:52 AM
  Subject: [SIPForum-discussion] Why Ack is different transaction?


  Hi,

  I want to know why ACK is considered different transaction from Invite transaction .

  regs


------------------------------------------------------------------------------


  _______________________________________________
  This is the SIP Forum discussion mailing list
  TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
  Post to the list at discussion at sipforum.org
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From sakcahalit at hotmail.com  Wed Mar 28 07:20:51 2007
From: sakcahalit at hotmail.com (Halit Sakca)
Date: Wed, 28 Mar 2007 10:20:51 +0300
Subject: [SIPForum-discussion] Authentication and authorization in SIP
Message-ID: <BAY126-W670CB6023CC4F864D6932CC6D0@phx.gbl>

Hi Sonja,Yes multiple applications can use same AA parameters, if you ask how, let me say that these parameters can be defined on a instance of DB server, so during the call handling, AA mechanism can  respond  to  multiple application.Thats my comment but I dont know that we are talking about same subject :)HalitDate: Tue, 27 Mar 2007 08:05:23 -0700From: belic_sonja at yahoo.comTo: discussion at sipforum.orgSubject: [SIPForum-discussion] Authentication and authorization in SIPHi, I have a question regarding authentication and authorization mechanism in SIP. For instance, if there are more then one applications running on the same SIP system, does every application authenticate itself or all of them use the same authentication parameters, defined for that SIP system? Thanks in advance.  Best Regards, Sonja 

No need to miss a message. Get email on-the-go with Yahoo! Mail for Mobile. Get started.
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From gowshan at sltnet.lk  Wed Mar 28 06:58:28 2007
From: gowshan at sltnet.lk (shankar)
Date: Wed, 28 Mar 2007 12:58:28 +0600
Subject: [SIPForum-discussion] Web based sip client
In-Reply-To: <00b601c770fe$23466ad0$8178a40a@china.huawei.com>
Message-ID: <0JFL006DOPYNQWB0@pop3.sltnet.lk>


Hi 

Please anyone give me free web based sip client to put in my web page to
communicate with my sip server

Please provide program to put in my web server 


Thanks

Regards
Shankar



From alexzhang at gdnt.com.cn  Wed Mar 28 07:48:32 2007
From: alexzhang at gdnt.com.cn (alexzhang at gdnt.com.cn)
Date: Wed, 28 Mar 2007 15:48:32 +0800
Subject: [SIPForum-discussion]
	=?gb2312?B?UkU6IFtTSVBGb3J1bS1kaXNjdXNzaW9uXVJlOiBbU0lQRm9ydW0tZA==?=	=?gb2312?B?aXNjdXNzaW9uXbTwuLQ6IGRpc2N1c3NpbwluIERpZ2VzdCwgVm9sIDIwLCA=?=	=?gb2312?B?SXNzdWUgMzg=?=
In-Reply-To: <00b601c770fe$23466ad0$8178a40a@china.huawei.com>
References: <OF0ECD4452.9BFB2601-ON482572AC.00137279@byd.com.cn>
	<00b601c770fe$23466ad0$8178a40a@china.huawei.com>
Message-ID: <8E523FC208B8174790E69947E307914701770975@rnd-ex01.rnd.gdnt.local>

Anybody in this list are involved in the development of the SIP-I (SIP w/ecanpsulated ISUP) ?


- A L E X

-----Original Message-----
From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Deepanshu
Sent: Wednesday, March 28, 2007 1:59 PM
To: wangran; discussion at sipforum.org
Subject: [SIPForum-discussion]Re: [SIPForum-discussion]??: discussio n Digest, Vol 20, Issue 38

Dear Wang

I don't think this create any kind of problems. 100 trying is hop-by-hop, the proxy can perform it simultaneously with other request (INVITE P1--->P2 in your case)

HTH

Deepanshu Gautam
R&D Engineer
Huawei Technologies Co. Ltd.
Nanjing, PRC

----- Original Message -----
From: "wangran" <wang.ran at byd.com.cn>
To: <discussion at sipforum.org>
Sent: Wednesday, March 28, 2007 11:31 AM
Subject: [SIPForum-discussion]??: discussion Digest, Vol 20, Issue 38


> Hi..
>          We have some question in RFC3261, May I have you some minute 
> to explain it?
>          Alice and bob?s call flaw in chapter 4 figure.1 is like this:
>
>          SIP Flow:
>
> -------------> INVITE (F1)\
>
> -------------> INVITE (F2)
>
> <------------- 100 Trying (F3)
>
>
>
>          But in chapter 24.2
>
>          F2 and F3 exchange there sequence
>
> -------------> INVITE (F1)\
>
> <------------- 100 Trying (F2)
>
> -------------> INVITE (F3)
>
>
>
> Does this small difference cause problems?
>
> your comment will be highly appreciated!
>
>
>
>
> Best of Regards,
>
> wangran
>
>  
> **********************************************************************
> *
> BYD TECHFAITH?COMPANY?LIMITED(BTC)
>
> Address:3/F,M8 West,NO.1 Jiu Xian Qiao Dong Road,
>         Chao Yang District,Beijing,China  100016
> PostCode:10016
> Mobile: +86-13810362150
> Tel: +86-10-58291226
> Mail: wang.ran at byd.com.cn
>  
> **********************************************************************
> *
>
>
> -----????-----
> ???: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.
> org] ?? discussion-request at sipforum.org
> ????: 2007?3?28? 0:00
> ???: discussion at sipforum.org
> ??: discussion Digest, Vol 20, Issue 38
>
> Send discussion mailing list submissions to discussion at sipforum.org
>
> To subscribe or unsubscribe via the World Wide Web, visit 
> http://sipforum.org/mailman/listinfo/discussion
> or, via email, send a message with subject or body 'help' to 
> discussion-request at sipforum.org
>
> You can reach the person managing the list at 
> discussion-owner at sipforum.org
>
> When replying, please edit your Subject line so it is more specific 
> than "Re: Contents of discussion digest..."
>
>
> Today's Topics:
>
>    1. simultaneous INVITEs (Niklas Fondberg)
>    2. R-Factor type equation to evaluate VOIP quality from
>       Wireshark RTP stats (Adam Harding)
>    3. Re: simultaneous INVITEs (Deepanshu)
>    4. Authentication and authorization in SIP (Sonja Belic)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 26 Mar 2007 21:30:24 +0200
> From: Niklas Fondberg <niklas.fondberg at tilgin.com>
> Subject: [SIPForum-discussion] simultaneous INVITEs
> To: discussion at sipforum.org
> Message-ID: <1174937424.5519.12.camel at localhost.localdomain>
> Content-Type: text/plain
>
> Hi,
> I new to this list but I hope that the list is what I'm after; an 
> implementation and design discussion list about SIP.
> If my question is wrongly addressed, please forgive me and please 
> point me the right direction...
>
> My question that I have searched all over for an answer to is quite
> simple:
>
> What is the correct behavior for a UA if a second INVITE arrives 
> before the first has been answered?
>
> SIP Flow:
>
> -------------> INVITE (1)
> <------------- 100 Trying (1)
> <------------- 180 Ringing (1)
> -------------> INVITE (2)
> ... ???
>
> Here the first (1) INVITE could have been answered by some other UA 
> that the INVITE might have been forked to and in this case session (2) 
> should start ringing.
>
>
> Niklas Fondberg
>
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Mon, 26 Mar 2007 23:01:13 +0100
> From: Adam Harding <adam.harding2 at ntlworld.com>
> Subject: [SIPForum-discussion] R-Factor type equation to evaluate VOIP 
> quality from Wireshark RTP stats
> To: "discussion at sipforum.org" <discussion at sipforum.org>
> Message-ID:
>
>
<20070326220113.LRUQ17393.aamtaout02-winn.ispmail.ntl.com at smtp.ntlworld.com>
>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hi,
>
> I am interested in any free algorithms that can be used to give a 
> value
for
> the voice quality in a VOIP call based on basic statistics such as 
> delay, packet loss and jitter.
>
> I asked this question a few days and got a really useful document 
> recommended to me which helps me understand how the R-Factor works but 
> I
can
> not get hold of the ITU-G values and my RTP results from Wireshark are 
> probably to basic to calculate the R-Factor.
>
> So just wondering if there is some sort of basic algorithm that I can
enter
> my results from the Wireshark RTP stats and get some sort of value of
voice
> quality that I can use to compare my results with each other.
>
> Thanks,
>
> Adam Harding
>
> -----------------------------------------
> Email sent from www.virginmedia.com/email Virus-checked using 
> McAfee(R) Software and scanned for spam
>
>
>
> ------------------------------
>
> Message: 3
> Date: Tue, 27 Mar 2007 09:50:17 +0800
> From: Deepanshu <deepanshu at huawei.com>
> Subject: Re: [SIPForum-discussion] simultaneous INVITEs
> To: Niklas Fondberg <niklas.fondberg at tilgin.com>
> Cc: discussion at sipforum.org
> Message-ID: <003801c77012$45a10050$8178a40a at china.huawei.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> inline
> ----- Original Message -----
> From: "Niklas Fondberg" <niklas.fondberg at tilgin.com>
> To: <discussion at sipforum.org>
> Sent: Tuesday, March 27, 2007 3:30 AM
> Subject: [SIPForum-discussion] simultaneous INVITEs
>
>
> > Hi,
> > I new to this list but I hope that the list is what I'm after; an 
> > implementation and design discussion list about SIP.
> > If my question is wrongly addressed, please forgive me and please 
> > point me the right direction...
> >
> > My question that I have searched all over for an answer to is quite
> > simple:
> >
> > What is the correct behavior for a UA if a second INVITE arrives 
> > before the first has been answered?
> >
> > SIP Flow:
> >
> > -------------> INVITE (1)
> > <------------- 100 Trying (1)
> > <------------- 180 Ringing (1)
> > -------------> INVITE (2)
> > ... ???
> >
> > Here the first (1) INVITE could have been answered by some other UA 
> > that the INVITE might have been forked to and in this case session 
> > (2) should start ringing.
>
> If the first (1) INVITE could have been answered by some other UA then 
> the originating UAC SHALL send a CANCEL request towards UAS instead of 
> a
INVITE
> (2).
>
> -------------> INVITE (1)
> <------------- 100 Trying (1)
> <------------- 180 Ringing (1)
> --------------> CANCEL (1)
>                         stop ringing
> <------------- SIP 487 (1)
>
>
> HTH
>
> Deepanshu Gautam
> R&D Engineer
> Huawei Technologies Co. Ltd.
> Nanjing, PRC
>
> >
> >
> > Niklas Fondberg
> >
> >
> >
> > _______________________________________________
> > This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or 
> > edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> > Post to the list at discussion at sipforum.org
> >
>
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 27 Mar 2007 08:05:23 -0700 (PDT)
> From: Sonja Belic <belic_sonja at yahoo.com>
> Subject: [SIPForum-discussion] Authentication and authorization in SIP
> To: discussion at sipforum.org
> Message-ID: <664094.34155.qm at web60623.mail.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>  I have a question regarding authentication and authorization 
> mechanism in SIP. For instance, if there are more then one 
> applications running on the same SIP system, does every application 
> authenticate itself or all of them use the same authentication parameters, defined for that SIP system?
>  Thanks in advance.
>
>  Best Regards,
>  Sonja
>
> ---------------------------------
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> your delivery options, please visit
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> org/mailman/listinfo/discussion
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From sukerry at 126.com  Wed Mar 28 09:54:30 2007
From: sukerry at 126.com (sukerry)
Date: Wed, 28 Mar 2007 17:54:30 +0800
Subject: [SIPForum-discussion] Authentication and authorization in SIP
Message-ID: <460A3B9A.034C77.20049@m5-143.126.com>

Sonja Belic???? 

??Every application authenticate itself

======== 2007-03-27 23:05:23 ???????? ========

Hi,
I have a question regarding authentication and authorization mechanism in SIP. For instance, if there are more then one applications running on the same SIP system, does every application authenticate itself or all of them use the same authentication parameters, defined for that SIP system?
Thanks in advance.

Best Regards,
Sonja


No need to miss a message. Get email on-the-go 
with Yahoo! Mail for Mobile. Get started. 

= = = = = = = = = = = = = = = = = = = = = = 
?????????
??

??????????????sukerry
??????????????sukerry at 126.com
???????????????2007-03-28
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From belic_sonja at yahoo.com  Wed Mar 28 10:01:02 2007
From: belic_sonja at yahoo.com (Sonja Belic)
Date: Wed, 28 Mar 2007 03:01:02 -0700 (PDT)
Subject: [SIPForum-discussion] Authentication and authorization in SIP
In-Reply-To: <664094.34155.qm@web60623.mail.yahoo.com>
Message-ID: <288514.96657.qm@web60616.mail.yahoo.com>

Hi, 
I'll try to explain my question in more details. If we have multihoming  ( more then one IP interface ) SIP system, with more then one client  applications, all implementing 
different services, and independently running on that SIP system, how 
should AA mechanism work? For instance, we use Digest auth scheme and 
receive challenge. The question is how shall we perform authentication, 
i.e. on which level? 
1. Should we have separate instance of Digest auth scheme for each 
application where each application controls own authentication parameters? 
2. Or we should consider just one Digest instance and set of 
authentication parameters performing authentication for system in whole 
(all applications shall use same, for example nonce, within their 
further SIP messages)? 
3. Or maybe we should take care about authentication per each interface 
of multihoming SIP system? 
 
This issue rise up the question about registration too. Shall we perform 
registration (SIP REGISTER request) for overall system at once or it 
should be done for each application separately? 
 
Are there any recommendations related to this issues? Is there any  dependency on the type of the network used? I would appreciate 
to get information about authentication practice in current SIP solutions. 
 
Thanks in advance. 
 
Best Regards, 
Sonja 

Sonja Belic <belic_sonja at yahoo.com> wrote: Hi,
 I have a question regarding authentication and authorization mechanism in SIP. For instance, if there are more then one applications running on the same SIP system, does every application authenticate itself or all of them use the same authentication parameters, defined for that SIP system?
 Thanks in advance.
 
 Best Regards,
 Sonja   

---------------------------------
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with Yahoo! Mail for Mobile. Get started._______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org


 
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From rakesh_rcm at yahoo.com  Wed Mar 28 11:03:23 2007
From: rakesh_rcm at yahoo.com (rakesh menon)
Date: Wed, 28 Mar 2007 04:03:23 -0700 (PDT)
Subject: [SIPForum-discussion] One way speech
Message-ID: <256511.56654.qm@web56607.mail.re3.yahoo.com>

Hi all,

has anyone come accross "one way speech" issue.
mostly happens to external incomming calls.
PSTN to IP Phone.
Is it something to do with drops in RTP packets.

Regards,
Rakesh


 
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From gkittu at gmail.com  Wed Mar 28 12:34:54 2007
From: gkittu at gmail.com (Krishna Kishore G)
Date: Wed, 28 Mar 2007 18:04:54 +0530
Subject: [SIPForum-discussion] One way speech
In-Reply-To: <256511.56654.qm@web56607.mail.re3.yahoo.com>
References: <256511.56654.qm@web56607.mail.re3.yahoo.com>
Message-ID: <49f7a1c10703280534l6aada2a5w94e1d0334a452008@mail.gmail.com>

Hi  Rakesh,
       I am assuming this because reachability of IP(of assaigned to
IP phone) from mediagateway.From mediagw(assuming the debugging tool
prescence) try to ping the IP address of phone.
Aside  you can take ethereal traces at the gateway level, or at access
point to check out wether RTP stream is flowing out from mediagw

Regards
Kishore

On 3/28/07, rakesh menon <rakesh_rcm at yahoo.com> wrote:
> Hi all,
>
> has anyone come accross "one way speech" issue.
> mostly happens to external incomming calls.
> PSTN to IP Phone.
> Is it something to do with drops in RTP packets.
>
> Regards,
> Rakesh
>
>
>
> ____________________________________________________________________________________
> No need to miss a message. Get email on-the-go
> with Yahoo! Mail for Mobile. Get started.
> http://mobile.yahoo.com/mail
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>


-- 
Thanks &regards
G.Krishna Kishore


From rjsparks at nostrum.com  Wed Mar 28 14:23:37 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Wed, 28 Mar 2007 09:23:37 -0500
Subject: [SIPForum-discussion] Why Ack is different transaction?
In-Reply-To: <0fd201c77103$92b192a0$6c19320a@telxsi.com>
References: <63af059d0703272122l5134afpa51426b1551dd36@mail.gmail.com>
	<0fd201c77103$92b192a0$6c19320a@telxsi.com>
Message-ID: <B477B5F6-055D-47DF-A2C2-7CFEA9E3F65D@nostrum.com>

The destruction of the INVITE transaction on 200 is a known bug in  
the spec and a correction will be published shortly
(if you actually destroy the transaction, you will treat  
retransmissions a new request).

The real reason an ACK-200 is a new transaction is that it must  
follow the route-set established by the 200 to the INVITE.
In other words, it may go to a completely different first-hop  
destination than the INVITE did. Thus, it needs its own transaction
identifier.

RjS

On Mar 28, 2007, at 1:37 AM, Avishek Chowdhury wrote:

> Hi Aditi,
>
> When the UAC receives 200 OK, the client transaction is destroyed  
> at TL.
> This is done because, the TL is unaware of the above core. The  
> above core can be a Proxy or an UAC core.
> In case of proxy, the 200 OK is forwarded whereas in case of UAC,  
> an ACK is generated. Now this ACK has to create a new transaction  
> (transaction created by INVITE had been destroyed)
> at TL for its transmission, hence the ACK for 200 OK is outside the  
> INVITE transaction.
>
> For other non-2xx final responses, the client transaction at TL is  
> not destroyed and the ACK is generated by TL. Hence in this case,  
> the ACK is within the transaction.
>
> Regards,
> Avishek
>
> ----- Original Message -----
> From: aditi g
> To: discussion at sipforum.org
> Sent: Wednesday, March 28, 2007 9:52 AM
> Subject: [SIPForum-discussion] Why Ack is different transaction?
>
> Hi,
>
> I want to know why ACK is considered different transaction from  
> Invite transaction .
>
> regs
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit http:// 
> sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit http:// 
> sipforum.org/mailman/listinfo/discussion
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From rjsparks at nostrum.com  Wed Mar 28 14:48:40 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Wed, 28 Mar 2007 09:48:40 -0500
Subject: [SIPForum-discussion] SIPit 20 registration closes in 2 days
Message-ID: <160B4420-F4CB-441E-96D1-EEA960D20724@nostrum.com>

SIPit 20 registration closes this Friday, March 30 (2 days from today).

If you are planning to attend, but have not registered, please do so  
now.

The details for the event are available at www.sipit.net

You can register using this link: https://www.regonline.com/? 
eventID=123004&rTypeID=89030

See you in Antwerp!

RjS


From qt.kiran at gmail.com  Wed Mar 28 14:56:54 2007
From: qt.kiran at gmail.com (kiran chakkilam)
Date: Wed, 28 Mar 2007 20:26:54 +0530
Subject: [SIPForum-discussion] hi
Message-ID: <e0646ce70703280756j1155ac60i201d089e0e08e869@mail.gmail.com>

Hi all,

I have doubts on basic registration
                        UA                          Registrarserver
                      Register---------------->
                                 < --------------------200 Ok

1)In this Scenario whether Register request & 200 Ok called as a transaction
or not?

2)Whether it's possilble to send the Register request with out branch
parameter in VIA header?

3)are there any possiblities to send the register request with out VIA
header n Registration?

4)If i send aRequire  header contains INVITE , CANCEL in the Register
request to Registrar server(DUT) whether it responds with unsupported
parameter or not

5) If i am sending INVITE request toward Registrar (DUT) what are the
expected responses.



Thanks in Advance


Ch.kiran
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From nirk at MICROSOFT.com  Wed Mar 28 15:37:30 2007
From: nirk at MICROSOFT.com (Nir Katz)
Date: Wed, 28 Mar 2007 16:37:30 +0100
Subject: [SIPForum-discussion] SIP messages route
Message-ID: <59DD872D2D837D44B60E6B6C630CE4B2142403EF0F@EA-EXMSG-C303.europe.corp.microsoft.com>

How common is the scenario where the UAC and UAS sends SIP messages to one another directly after they have established the dialog and started media exchange? Or do most UAs continue to send the SIP messages using the same route they came from?   Does the protocol they use influence this behavior?

Thanks in advance

Nir Katz
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From indresh.singh at siemens.com  Wed Mar 28 18:05:39 2007
From: indresh.singh at siemens.com (Singh, Indresh (SNL US))
Date: Wed, 28 Mar 2007 11:05:39 -0700
Subject: [SIPForum-discussion] hi
In-Reply-To: <e0646ce70703280756j1155ac60i201d089e0e08e869@mail.gmail.com>
Message-ID: <3D80B10873C01D47BEC71C8DE311CF111CE07AC4@USNWK100MSX.ww017.siemens.net>

Below is my understanding. Hopefully it would help.
 
Regards,
 
Indresh

________________________________

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of kiran chakkilam
Sent: Wednesday, March 28, 2007 10:57 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] hi


Hi all,
 
I have doubts on basic registration
                        UA                          Registrarserver
                      Register---------------->
                                 < --------------------200 Ok
 
1)In this Scenario whether Register request & 200 Ok called as a
transaction or not?
[Singh, Indresh] Yes for devices compatible with RFC3261. 
 
2)Whether it's possilble to send the Register request with out branch
parameter in VIA header?
[Singh, Indresh] It is possible to send register request without branch
parameter in via header (if SIP device is compliant only with
RFC-2543/previous SIP RFC ). In that case the registrar server has to be
backward compatible with RFC-2543 to be able to process this request.
RFC-3261 recommends that SIP servers should be able to process requests
without branch parameter to maintain backward compatibility with
RFC-2543 
 
3)are there any possiblities to send the register request with out VIA
header n Registration?
[Singh, Indresh] No. Via header is mandatory in the requests. Refer to
Table 3 on page 163 of RFC-3261 Without via header in requests the
responses can not be sent. 
 
4)If i send aRequire  header contains INVITE , CANCEL in the Register
request to Registrar server(DUT) whether it responds with unsupported
parameter or not
[Singh, Indresh] Require Header or Allow header. Require header has tags
like timer ( indicating session timer support ) 100 rel ( indicating
PRACK support ) ?? 
 
5) If i am sending INVITE request toward Registrar (DUT) what are the
expected responses.
 Logically 405 Method Not allowed. Page 186
 
 
Thanks in Advance
 
 
Ch.kiran
 
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From parrishsteve2000 at yahoo.com  Wed Mar 28 18:58:05 2007
From: parrishsteve2000 at yahoo.com (Steve Parrish)
Date: Wed, 28 Mar 2007 13:58:05 -0500
Subject: [SIPForum-discussion] SIP messages route
In-Reply-To: <59DD872D2D837D44B60E6B6C630CE4B2142403EF0F@EA-EXMSG-C303.europe.corp.microsoft.com>
Message-ID: <008101c7716b$0660b930$031410ac@ibmhdqj6pzqq1l>

If you're talking about SIP endpoints that span subnets I would say it's
not very common at all due to NAT/firewall traversal.  Also with the
increasing number of SBC's (Session Border Controllers) being deployed I
would say that Service Providers force SIP traffic to hop along these
nodes.
 
Does the protocol they use influence this behavior?
Not really, there's no guarantee since the connection could be part UDP
and part TCP.
 
-Steve P.
 
 
-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Nir Katz
Sent: Wednesday, March 28, 2007 10:38 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] SIP messages route
 
How common is the scenario where the UAC and UAS sends SIP messages to
one another directly after they have established the dialog and started
media exchange? Or do most UAs continue to send the SIP messages using
the same route they came from?   Does the protocol they use influence
this behavior?
 
Thanks in advance
 
Nir Katz
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From mohandivakar2005 at yahoo.co.in  Thu Mar 29 07:43:27 2007
From: mohandivakar2005 at yahoo.co.in (mohan divakar)
Date: Thu, 29 Mar 2007 08:43:27 +0100 (BST)
Subject: [SIPForum-discussion] query regarding rtp loss in sip end points
Message-ID: <624344.55611.qm@web8606.mail.in.yahoo.com>

Hi,
  I have a question regarding rtp loss,
  there is a connection established between two sip users but both cant hear the voice of each other.
  I just want to know the reason why there is a loss of rtp when the connection is already established.
   
  thanks in advance
  Mohan
   

 				
---------------------------------
 Here?s a new way to find what you're looking for - Yahoo! Answers 
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From deepak.k at globaledgesoft.com  Thu Mar 29 11:09:47 2007
From: deepak.k at globaledgesoft.com (Deepak K)
Date: Thu, 29 Mar 2007 16:39:47 +0530
Subject: [SIPForum-discussion] SIPit 20 : what's about IMS Interop?
References: <3D80B10873C01D47BEC71C8DE311CF111CE07AC4@USNWK100MSX.ww017.siemens.net>
Message-ID: <00c701c771f2$c3727130$520710ac@globaledgesoft.com>

HI,

SIP is a major element of the IMS architecture.

Wanted to know whether IMS/3GPP specific SIP extension/implementations interop would be one of the major focus point of this event ? 

Regards,
DK
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From sipcbi at yahoo.co.in  Thu Mar 29 12:42:09 2007
From: sipcbi at yahoo.co.in (sip cbi)
Date: Thu, 29 Mar 2007 13:42:09 +0100 (BST)
Subject: [SIPForum-discussion] can use jsr 281  for creating SIP phone??
Message-ID: <957311.22569.qm@web94312.mail.in2.yahoo.com>

Dear All,
   
  Greetings!!!
   
  can we create SIP Phone (User Agent) using JSR 281 through Java 2 standard Edition ?
   
  Thanks,
  sipcbi

 				
---------------------------------
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From oguzhan at nevotek.com  Thu Mar 29 13:12:16 2007
From: oguzhan at nevotek.com (Oguzhan Cem)
Date: Thu, 29 Mar 2007 16:12:16 +0300
Subject: [SIPForum-discussion] can use jsr 281 for creating SIP phone??
Message-ID: <6F0695B2993AAF4EB5A63AEA95A44BBFF9386A@buddy.nevotek.com>

Hello,

 

I also want to ask a question related,

 

If we create a sip agent, (with any platform including JSR 281) does
anyone give me any idea how to test it? Any test site installed for this
purpose?

 

Thx in advance,

 

Oguzhan Cem.

 

 

 

  _____  

From: sip cbi [mailto:sipcbi at yahoo.co.in] 
Sent: Thursday, March 29, 2007 3:42 PM
To: 
Subject: [SIPForum-discussion] can use jsr 281 for creating SIP phone??

 

Dear All,

 

Greetings!!!

 

can we create SIP Phone (User Agent) using JSR 281 through Java 2
standard Edition ?

 

Thanks,

sipcbi

  

  _____  

Here's a new way to find what you're looking for - Yahoo! Answers
<http://us.rd.yahoo.com/mail/in/yanswers/*http:/in.answers.yahoo.com/>  

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From indresh.singh at siemens.com  Thu Mar 29 20:41:49 2007
From: indresh.singh at siemens.com (Singh, Indresh (SNL US))
Date: Thu, 29 Mar 2007 13:41:49 -0700
Subject: [SIPForum-discussion] query regarding rtp loss in sip end points
In-Reply-To: <624344.55611.qm@web8606.mail.in.yahoo.com>
Message-ID: <3D80B10873C01D47BEC71C8DE311CF111D1CA179@USNWK100MSX.ww017.siemens.net>

SIP does only the signaling and during the signaling the SDP/media is
exchanged between the two devices using SIP Signaling protocol. If the
SDP exchanged between two devices do not properly exchange
 
1) codecs
2) RTP port and addresses where they will send and receive media
 
You will not have a speech path. This could be one reason why you do not
have speec path. There could be other reasons as well....
 
Regards,
 
Indresh K Singh

________________________________

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of mohan divakar
Sent: Thursday, March 29, 2007 3:43 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] query regarding rtp loss in sip end
points


Hi,
I have a question regarding rtp loss,
there is a connection established between two sip users but both cant
hear the voice of each other.
I just want to know the reason why there is a loss of rtp when the
connection is already established.
 
thanks in advance
Mohan
 

________________________________

Here's a new way to find what you're looking for - Yahoo! Answers
<http://us.rd.yahoo.com/mail/in/yanswers/*http://in.answers.yahoo.com/>

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From nvvgopal80 at rediffmail.com  Fri Mar 30 02:43:19 2007
From: nvvgopal80 at rediffmail.com (venkata venu gopal)
Date: 30 Mar 2007 02:43:19 -0000
Subject: [SIPForum-discussion] can use jsr 281 for creating SIP phone??
Message-ID: <20070330024319.1675.qmail@webmail99.rediffmail.com>

Hello,

There are test frameworks for the SIP to test the device for the confirmance of the RFCs, some torture test cases and interoperability and etc. Find below the URLs to find the test framework and the test suites.. 

Test Framework : http://www.sipfoundry.org/sip-forum-test-framework/sip-forum-test-framework-sftf.html

Test Suites :  http://www.iol.unh.edu/services/testing/voip/testsuites/#SIP%20Conformance%20Test%20Suite

There may be other sites better then this but any information in this direction are welcomed and appreciated.

Hope this is helpful to you.. 

Thanks,
Venu. ?


On Thu, 29 Mar 2007 Oguzhan Cem wrote :
>Hello,
>
>
>
>I also want to ask a question related,
>
>
>
>If we create a sip agent, (with any platform including JSR 281) does
>anyone give me any idea how to test it? Any test site installed for this
>purpose?
>
>
>
>Thx in advance,
>
>
>
>Oguzhan Cem.
>
>
>
>
>
>
>
>   _____
>
> From: sip cbi [mailto:sipcbi at yahoo.co.in]
>Sent: Thursday, March 29, 2007 3:42 PM
>To:
>Subject: [SIPForum-discussion] can use jsr 281 for creating SIP phone??
>
>
>
>Dear All,
>
>
>
>Greetings!!!
>
>
>
>can we create SIP Phone (User Agent) using JSR 281 through Java 2
>standard Edition ?
>
>
>
>Thanks,
>
>sipcbi
>
>
>
>   _____
>
>Here's a new way to find what you're looking for - Yahoo! Answers
><http://us.rd.yahoo.com/mail/in/yanswers/*http:/in.answers.yahoo.com/>
>
>_______________________________________________
>This is the SIP Forum discussion mailing list
>TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
>Post to the list at discussion at sipforum.org
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From rjsparks at nostrum.com  Fri Mar 30 04:21:34 2007
From: rjsparks at nostrum.com (Robert Sparks)
Date: Thu, 29 Mar 2007 23:21:34 -0500
Subject: [SIPForum-discussion] SIPit 20 registration closes today (March 30)
Message-ID: <E2CAE4B2-6A2C-434B-BD09-5E5799300B04@nostrum.com>

Registration for SIPit 20 closes today (March 30).

If you plan to attend, but have not yet registered, do so immediately.

Information on the event and the link for registration can be found  
at www.sipit.net.

See you in Antwerp!

RjS


From invite at friends.unicefusa.org  Fri Mar 30 10:09:45 2007
From: invite at friends.unicefusa.org (gujjenaveen@gmail.com)
Date: Fri, 30 Mar 2007 03:09:45 -0700
Subject: [SIPForum-discussion] Inviting my friends & family...
Message-ID: <1175249385.27076@unicef.popularmediamail.org>

I'm extending a personal invitation to my friends and family to make a difference without spending a penny.  To see your invitation, click the link below, or copy and paste it into your browser's address field:

http://friends.unicefusa.org/r/6eaf7b3c2fdf102a8325











If you would prefer not to receive invitations from Friends.UNICEFUSA.org please click here 
http://friends.unicefusa.org/?PC=UNSUB&rh=22ce585c3ac9ab673813db0da6408e93&sender=naveeng at intoto.com&tc=12

----------------------------------------------------------
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PMB# 210
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San Francisco, CA 94103-4211



From hariprasad.taduru at gmail.com  Fri Mar 30 09:20:20 2007
From: hariprasad.taduru at gmail.com (Taduru Hariprasad)
Date: Fri, 30 Mar 2007 14:50:20 +0530
Subject: [SIPForum-discussion] Hi
Message-ID: <4dff78790703300220v4b08a07cm5b17f502ecbade89@mail.gmail.com>

Hi,

I started to learn sip. Currently going thru rfc 3261. Can i have th
answers for the
following doubts.

1) BRANCH and CSeq parameters will get change for every transaction?

2) How Ack is treated as one transaction if the final response is 200-ok for
   an INVITE?

3) How to detect loops and overcome them?

And also please mension/attach the docs if you have for my referrence.

Thanks
Hari


From sipcbi at yahoo.co.in  Fri Mar 30 10:45:55 2007
From: sipcbi at yahoo.co.in (sip cbi)
Date: Fri, 30 Mar 2007 11:45:55 +0100 (BST)
Subject: [SIPForum-discussion] how to create java installable ?????
Message-ID: <278653.85848.qm@web94304.mail.in2.yahoo.com>

Hello All !!!!

how to create java exe file and installable????
is there any tool to make java exe ....
please give me the site to download the tool....

Regards
sipcbi

 				
---------------------------------
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From indresh.singh at siemens.com  Fri Mar 30 14:58:11 2007
From: indresh.singh at siemens.com (Singh, Indresh (SNL US))
Date: Fri, 30 Mar 2007 07:58:11 -0700
Subject: [SIPForum-discussion] Hi
In-Reply-To: <4dff78790703300220v4b08a07cm5b17f502ecbade89@mail.gmail.com>
Message-ID: <3D80B10873C01D47BEC71C8DE311CF111D1CA47A@USNWK100MSX.ww017.siemens.net>

 

-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Taduru Hariprasad
Sent: Friday, March 30, 2007 5:20 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] Hi

Hi,

I started to learn sip. Currently going thru rfc 3261. Can i have th
answers for the
following doubts.

1) BRANCH and CSeq parameters will get change for every transaction?

>> Yes. 

2) How Ack is treated as one transaction if the final response is 200-ok
for
   an INVITE?

>> Yes. But it is a unique transaction and it's cseq is same as
INVITE-200OK. Just the branch-identifier is different.

3) How to detect loops and overcome them?

>> By checking if the ipAddress of your box is there in one of the i/c
via headers. But after this you have to differentiate between loop and
spiraling before you can say loop detected. 

And also please mension/attach the docs if you have for my referrence.

>> Everything in RFC-3261 :)

Thanks
Hari
_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org



From hsingh at wgate.com  Fri Mar 30 15:41:37 2007
From: hsingh at wgate.com (Harbinder Singh)
Date: Fri, 30 Mar 2007 11:41:37 -0400
Subject: [SIPForum-discussion] Hi
In-Reply-To: <3D80B10873C01D47BEC71C8DE311CF111D1CA47A@USNWK100MSX.ww017.siemens.net>
Message-ID: <FA7EECCA3735EF46BE49245D249B788CED6B25@exchange1.wgate.com>

Also, more explanation in line:--

-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Singh, Indresh
(SNL US)
Sent: Friday, March 30, 2007 10:58 AM
To: Taduru Hariprasad; discussion at sipforum.org
Subject: Re: [SIPForum-discussion] Hi

 

-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Taduru Hariprasad
Sent: Friday, March 30, 2007 5:20 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] Hi

Hi,

I started to learn sip. Currently going thru rfc 3261. Can i have th
answers for the
following doubts.

1) BRANCH and CSeq parameters will get change for every transaction?

>> Yes. 
----For example, in a call: Each successive request during a call will
have a higher CSeq number. Also, the caller and the called parties each
maintain their own separate Cseq counts.

2) How Ack is treated as one transaction if the final response is 200-ok
for
   an INVITE?

>> Yes. But it is a unique transaction and it's cseq is same as
INVITE-200OK. Just the branch-identifier is different.

---- INVITE is the only method in SIP in which there is this three-way
handshake involving ACK. All other SIP requests are of the form
REQUEST/200 or REQUEST/4xx or 5xx or 6xx for a failure. That is why ACK
has the same Cseq number as the other two - INVITE and 200.

3) How to detect loops and overcome them?

>> By checking if the ipAddress of your box is there in one of the i/c
via headers. But after this you have to differentiate between loop and
spiraling before you can say loop detected. 

And also please mension/attach the docs if you have for my referrence.

>> Everything in RFC-3261 :)

Thanks
Hari
_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org

_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org




From suryaprakashu at rediffmail.com  Fri Mar 30 16:07:49 2007
From: suryaprakashu at rediffmail.com (Surya  Prakash Ummadi)
Date: 30 Mar 2007 16:07:49 -0000
Subject: [SIPForum-discussion] Hi
Message-ID: <20070330160749.26122.qmail@webmail36.rediffmail.com>

 ?
Hi,

   please comment below  following  answers :


1) BRANCH and CSeq parameters will get change for every transaction?
>
> >> Yes.(this is correct)
>
>2) How Ack is treated as one transaction if the final response is 200-ok
>for
>    an INVITE?
>
> >> Yes. But it is a unique transaction and it's cseq is same as
>INVITE-200OK. Just the branch-identifier is different.
>
>3) How to detect loops and overcome them?
>
> >> For detecting loops ,branch-id is used as the reference.Branch-id is formed by the magiccooke and hash of the request-uri,from tag,totag ,cseq,callid and topmost VIA header.IF it is not the loop,atleast request-uri will change.So branch-id will be different.IF it same,then loop is detected.

Another way is MAX-Forwards.
>
>And also please mension/attach the docs if you have for my referrence.

  refer to the tech-invite site,where u can find good presentation of documents.
>
On Fri, 30 Mar 2007 Singh,Indresh(SNL US) wrote :
>
>

regards
surya
>-----Original Message-----
> From: discussion-bounces at sipforum.org
>[mailto:discussion-bounces at sipforum.org] On Behalf Of Taduru Hariprasad
>Sent: Friday, March 30, 2007 5:20 AM
>To: discussion at sipforum.org
>Subject: [SIPForum-discussion] Hi
>
>Hi,
>
>I started to learn sip. Currently going thru rfc 3261. Can i have th
>answers for the
>following doubts.
>
>1) BRANCH and CSeq parameters will get change for every transaction?
>
> >> Yes.
>
>2) How Ack is treated as one transaction if the final response is 200-ok
>for
>    an INVITE?
>
> >> Yes. But it is a unique transaction and it's cseq is same as
>INVITE-200OK. Just the branch-identifier is different.
>
>3) How to detect loops and overcome them?
>
> >> By checking if the ipAddress of your box is there in one of the i/c
>via headers. But after this you have to differentiate between loop and
>spiraling before you can say loop detected.
>
>And also please mension/attach the docs if you have for my referrence.
>
> >> Everything in RFC-3261 :)
>
>Thanks
>Hari
>_______________________________________________
>This is the SIP Forum discussion mailing list
>TO UNSUBSCRIBE, or edit your delivery options, please visit
>http://sipforum.org/mailman/listinfo/discussion
>Post to the list at discussion at sipforum.org
>
>_______________________________________________
>This is the SIP Forum discussion mailing list
>TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
>Post to the list at discussion at sipforum.org
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From tkishor at softlinkindia.com  Sat Mar 31 03:07:42 2007
From: tkishor at softlinkindia.com (Kishor)
Date: Sat, 31 Mar 2007 08:37:42 +0530
Subject: [SIPForum-discussion] (no subject)
Message-ID: <001601c77341$ea265e40$d728e0dc@qpo2pp>

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From am2866 at columbia.edu  Sat Mar 31 08:27:03 2007
From: am2866 at columbia.edu (Arpit Mehta)
Date: Sat, 31 Mar 2007 04:27:03 -0400
Subject: [SIPForum-discussion] Regarding Caller ID
Message-ID: <bbd5fc140703310127q4f5e643fhaae3a2566cf597bd@mail.gmail.com>

Hello,

I have a problem regarding Caller ID. I am running the SipC client on my
machine. I am using a Cisco 2600 router. I have configured the gateway so as
to connect to my SipC client when a call is made to the gateway. Now when I
make a call, it connects to the SipC and everything works fine. But I need
the number of the person who has called SipC. In SipC also it displays
unknown and does not display the number. Are there any possible reasons for
the number not being shown? Does it look to be a configuration problem at
the router side so that it is not getting the caller's ID?

Any suggestion would be helpful.

Thanks.

--
Arpit
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From umair3210 at yahoo.com  Sat Mar 31 12:07:33 2007
From: umair3210 at yahoo.com (Muhammad Umair)
Date: Sat, 31 Mar 2007 05:07:33 -0700 (PDT)
Subject: [SIPForum-discussion] programming SIP??? help me
Message-ID: <117401.15464.qm@web38711.mail.mud.yahoo.com>

hi all,
   
  i have done the reading about the SIP. can any one tell me how can programatically (using any languange C#.net,vb.net) implement SIP.
   
  thanx in advance
  Regards
  Muhammad Umair

 
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From jani_tech_forum at yahoo.com  Sat Mar 31 14:13:47 2007
From: jani_tech_forum at yahoo.com (Janakiraman N)
Date: Sat, 31 Mar 2007 07:13:47 -0700 (PDT)
Subject: [SIPForum-discussion] programming SIP??? help me
In-Reply-To: <117401.15464.qm@web38711.mail.mud.yahoo.com>
Message-ID: <562500.36371.qm@web62111.mail.re1.yahoo.com>

Hi Muhammad Umair,

You can implement your SIP knowledge using SIP Servlet which will be used to create a SIP service. Its basically JAVA.

If you want to know in detail, please read JSR 116 standard SIP Servlet

Regards,
Janakiraman. N

Muhammad Umair <umair3210 at yahoo.com> wrote: hi all,
   
  i have done the reading about the SIP. can any one tell me how can programatically (using any languange C#.net,vb.net) implement SIP.
   
  thanx in advance
  Regards
  Muhammad Umair
   

---------------------------------
No need to miss a message. Get email on-the-go 
with Yahoo! Mail for Mobile. Get started._______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org


 
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From karim_balkas at hotmail.com  Sat Mar 31 18:49:10 2007
From: karim_balkas at hotmail.com (balkas karim)
Date: Sat, 31 Mar 2007 18:49:10 +0000
Subject: [SIPForum-discussion] simulate sip with NS2
Message-ID: <BAY12-F22A1E34D54AB6DA3624EF8FA620@phx.gbl>

hi all,

if you can help me about simulation the protocol SIP with Network Simulator 
2 (NS2)!!!

thanks for response!!

karim

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