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WebRTC Support #24

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bklang opened this issue Jun 26, 2013 · 3 comments

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bklang commented Jun 26, 2013

Each telephony engine should support WebRTC (if possible)

Asterisk: https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
FreeSWITCH: http://lists.freeswitch.org/pipermail/freeswitch-users/2013-June/097030.html

The Adhearsion VM should contain a small server that includes a WebRTC demo app (SIPML5 or JsSIP demos would work).

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bklang commented Oct 9, 2014

Docs on configuring FreeSWITCH to support WebRTC: https://confluence.freeswitch.org/display/FREESWITCH/WebRTC

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bklang commented Oct 9, 2014

Step-by-step for configuring Asterisk to support WebRTC: http://sipjs.com/guides/server-configuration/asterisk/

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bklang commented Aug 3, 2015

See also #50

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