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

Wed Mar 28 05:58:41 UTC 2007

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
>
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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
>
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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
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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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