[asterisk-dev] [Code Review]: WebSocket SIP Support

[opticron]

> On July 9, 2012, 2 p.m., opticron wrote:
> > The only issue I see here is if 'avpf=yes' is set and Asterisk sends an outbound INVITE with AVPF/SAVPF.  When this happens, the other end may have modified the stream to use AVP/SAVP instead of AVPF/SAVPF, but Asterisk will still use AVPF/SAVPF (if there is actually a difference at this point).
> 
> Kevin Fleming wrote:
>     If Asterisk sends an SDP offer with a media stream using RTP/(S)AVPF, any SDP answer to that offer *must* specify the same profile; the answer cannot specify RTP/(S)AVP.

I thought that was the case, but I must have read the RFC incorrectly when I was reviewing this.  Given that, Asterisk needs to reject any incoming offers that don't match the 'avpf=yes|no' setting for that peer or else it will generate an invalid SDP response.


- opticron


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On July 9, 2012, 10:59 a.m., Joshua Colp wrote:
> 
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> 
> (Updated July 9, 2012, 10:59 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> These changes add WebSocket transport support to chan_sip and fix some minor issues uncovered in the stack when used with WebSocket as a transport.
> 
> 
> Diffs
> -----
> 
>   /trunk/channels/chan_sip.c 369768 
>   /trunk/channels/sip/include/sip.h 369768 
>   /trunk/channels/sip/sdp_crypto.c 369768 
>   /trunk/channels/sip/security_events.c 369768 
>   /trunk/configs/sip.conf.sample 369768 
>   /trunk/include/asterisk/http_websocket.h 369768 
>   /trunk/res/res_http_websocket.c 369768 
> 
> Diff: https://reviewboard.asterisk.org/r/2008/diff
> 
> 
> Testing
> -------
> 
> Tested using sipml5 javascript SIP stack. Confirmed protocol traffic is correct, that connections are shutdown properly when they should be, that registration works, and that calling works.
> 
> 
> Thanks,
> 
> Joshua
> 
>

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