[asterisk-dev] [Code Review] 3438: Implement SIP TImer C in Asterisk

[Olle E Johansson]

> On April 14, 2014, 1:44 a.m., Matt Jordan wrote:
> > I'm sure I'm missing something obvious, but in what scenario do we forward a request in a manner consistent with a proxy? I'm thinking of those scenarios where an inbound INVITE request is received by Asterisk, and something in the dialplan causes chan_sip to forward the INVITE request to something outside of Asterisk.

Please don't be confused that the RFC mentions this is for proxys. Bad things will happen if we get a 100 trying and then nothing - Asterisk will wait forever for a final response. You do not want that in your servers. The best possible solution is to implement Timer C, since it's defined.


- Olle E


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On April 11, 2014, 10:41 a.m., Olle E Johansson wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3438/
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> 
> (Updated April 11, 2014, 10:41 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Repository: Asterisk
> 
> 
> Description
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> 
> 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
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> 
>   /trunk/configs/sip.conf.sample 412166 
>   /trunk/channels/sip/include/sip.h 412166 
>   /trunk/channels/chan_sip.c 412166 
> 
> Diff: https://reviewboard.asterisk.org/r/3438/diff/
> 
> 
> Testing
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> 
> Passed interoperability testing with funky test tool.
> 
> 
> Thanks,
> 
> Olle E Johansson
> 
>

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