[SIPForum-discussion] 答复: discussion Digest, Vol 20, Issue 38

Wed Mar 28 03:31:43 UTC 2007

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

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发送时间: 2007年3月28日 0:00
收件人: discussion at sipforum.org
主题: discussion Digest, Vol 20, Issue 38

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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

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

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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