[asterisk-dev] [Code Review] 2493: Add WebSocket transport module
Jason Parker
reviewboard at asterisk.org
Wed May 8 11:13:04 CDT 2013
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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 8, 2013, 4:13 p.m.)
Review request for Asterisk Developers.
Changes
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Address feedback.
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 (updated)
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/team/group/pimp_my_sip/include/asterisk/res_sip.h 387967
/team/group/pimp_my_sip/res/res_sip.c 387967
/team/group/pimp_my_sip/res/res_sip.exports.in 387967
/team/group/pimp_my_sip/res/res_sip/config_transport.c 387967
/team/group/pimp_my_sip/res/res_sip/location.c 387967
/team/group/pimp_my_sip/res/res_sip_outbound_registration.c 387967
/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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