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

[Joshua Colp]

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Review request for Asterisk Developers.


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
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  /trunk/channels/chan_sip.c 369517 
  /trunk/channels/sip/include/sip.h 369516 
  /trunk/channels/sip/sdp_crypto.c 369516 
  /trunk/channels/sip/security_events.c 369516 
  /trunk/configs/sip.conf.sample 369516 

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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