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

[Joshua Colp]

> On July 11, 2012, 8:20 a.m., opticron wrote:
> > /trunk/channels/chan_sip.c, lines 9481-9485
> > <https://reviewboard.asterisk.org/r/2008/diff/3/?file=29999#file29999line9481>
> >
> >     This should also decline the offer when AVPF is enabled, but AVP/SAVP is received.

Fixed.


> On July 11, 2012, 8:20 a.m., opticron wrote:
> > /trunk/channels/chan_sip.c, lines 9544-9548
> > <https://reviewboard.asterisk.org/r/2008/diff/3/?file=29999#file29999line9544>
> >
> >     Ditto.

Fixed.


On July 11, 2012, 8:20 a.m., Joshua Colp wrote:
> > This should also be done for text streams.

Fixed.


- Joshua


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On July 11, 2012, 7:12 a.m., Joshua Colp wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2008/
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> 
> (Updated July 11, 2012, 7:12 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 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 
> 
> 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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