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

Wed Mar 28 06:23:21 UTC 2007

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]答复: 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)
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> >
> > -----邮件原件-----
> > 发件人: discussion-bounces at sipforum.org
> [mailto:discussion-bounces at sipforum.
> > org] 代表 discussion-request at sipforum.org
> > 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
> >
> > Send discussion mailing list submissions to
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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
> >
> > -----------------------------------------
> > 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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