[asterisk-dev] [Code Review] 2493: Add WebSocket transport module

[svnbot]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2493/
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(Updated May 17, 2013, 2:38 p.m.)


Status
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This change has been marked as submitted.


Review request for Asterisk Developers.


Changes
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Committed in revision 389004


Bugs: ASTERISK-20952
    https://issues.asterisk.org/jira/browse/ASTERISK-20952


Repository: Asterisk


Description
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Adds a custom WebSocket transport module.


Diffs
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  /team/group/pimp_my_sip/include/asterisk/res_sip.h 388617 
  /team/group/pimp_my_sip/res/res_sip.c 388617 
  /team/group/pimp_my_sip/res/res_sip.exports.in 388617 
  /team/group/pimp_my_sip/res/res_sip/config_transport.c 388617 
  /team/group/pimp_my_sip/res/res_sip/include/res_sip_private.h 388617 
  /team/group/pimp_my_sip/res/res_sip/location.c 388617 
  /team/group/pimp_my_sip/res/res_sip_outbound_registration.c 388617 
  /team/group/pimp_my_sip/res/res_sip_transport_websocket.c PRE-CREATION 

Diff: https://reviewboard.asterisk.org/r/2493/diff/


Testing
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Registration works, incoming (from the browser) and outgoing (to the browser) calls work, audio flows (though, I don't have a mic, so I didn't test audio that direction).


Thanks,

Jason Parker

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