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

[opticron]

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/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/2008/#comment12656>

    This should also decline the offer when AVPF is enabled, but AVP/SAVP is received.



/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/2008/#comment12657>

    Ditto.


This should also be done for text streams.

- opticron


On July 11, 2012, 7:12 a.m., Joshua Colp wrote:
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> (Updated July 11, 2012, 7:12 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 369847 
>   /trunk/channels/sip/include/sip.h 369836 
>   /trunk/channels/sip/sdp_crypto.c 369836 
>   /trunk/channels/sip/security_events.c 369836 
>   /trunk/configs/sip.conf.sample 369836 
>   /trunk/include/asterisk/http_websocket.h 369836 
>   /trunk/res/res_http_websocket.c 369836 
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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,
> 
> Joshua
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>

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