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

[Jason Parker]

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/team/group/pimp_my_sip/res/res_sip.c
<https://reviewboard.asterisk.org/r/2493/#comment16521>

    This is supposed to be &&.  Already fixed.


- Jason Parker


On May 8, 2013, 4:13 p.m., Jason Parker wrote:
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> https://reviewboard.asterisk.org/r/2493/
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> (Updated May 8, 2013, 4:13 p.m.)
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> Review request for Asterisk Developers.
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> Bugs: ASTERISK-20952
>     https://issues.asterisk.org/jira/browse/ASTERISK-20952
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> Repository: Asterisk
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> Description
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> Adds a custom WebSocket transport module.
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> Diffs
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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 
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> Diff: https://reviewboard.asterisk.org/r/2493/diff/
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> 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).
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> Thanks,
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> Jason Parker
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