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

[opticron]

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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).

- opticron


On July 9, 2012, 10:59 a.m., Joshua Colp wrote:
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> (Updated July 9, 2012, 10:59 a.m.)
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> 
> Review request for Asterisk Developers.
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> Summary
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> 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.
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> Diffs
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>   /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 
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> Diff: https://reviewboard.asterisk.org/r/2008/diff
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> Testing
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> 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.
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> Thanks,
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> Joshua
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>

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