[asterisk-dev] [Code Review] 3438: Implement SIP TImer C in Asterisk
Matt Jordan
reviewboard at asterisk.org
Wed May 21 07:53:35 CDT 2014
> On April 23, 2014, 9:13 a.m., Mark Michelson wrote:
> > /trunk/channels/chan_sip.c, lines 22992-22996
> > <https://reviewboard.asterisk.org/r/3438/diff/1/?file=57236#file57236line22992>
> >
> > RFC 3261 section 16.7, bullet point 2 states:
> >
> > "...if the response is a provisional response with status codes 101 to 199 inclusive (i.e., anything but 100), the proxy MUST reset timer C..."
> >
> > This else case needs to be modified to not execute if the response is a 100.
I think I agree with Mark here as well, and the RFC makes sense when it says to not reset Timer C on a 100. There's no real need to reset Timer C in that case, as that typically comes directly on the heels of the transmitted INVITE request.
- Matt
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On May 16, 2014, 7:32 a.m., Olle E Johansson wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3438/
> -----------------------------------------------------------
>
> (Updated May 16, 2014, 7:32 a.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> SIP Timer C is defined for proxys that forward messages. In some ways, we forward calls. It is activated when we receive a 100 trying and wait for any other message. If that's not received, timer C triggers and cancels the call attempt.
>
> This is required in an interoperability test I'm working with.
>
> Red dots will be handled in the way they deserve.
>
>
> Diffs
> -----
>
> /trunk/configs/sip.conf.sample 414046
> /trunk/channels/sip/include/sip.h 414046
> /trunk/channels/chan_sip.c 414046
> /trunk/CHANGES 414046
>
> Diff: https://reviewboard.asterisk.org/r/3438/diff/
>
>
> Testing
> -------
>
> Passed interoperability testing with funky test tool.
>
>
> Thanks,
>
> Olle E Johansson
>
>
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