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

[Joshua Colp]

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(Updated July 11, 2012, 8:37 a.m.)


Review request for Asterisk Developers.


Changes
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More rejection tweaks. Those bad media streams should feel oh so rejected.


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.


Diffs (updated)
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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 

Diff: https://reviewboard.asterisk.org/r/2008/diff


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.


Thanks,

Joshua

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