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adhearsion / Telephony-Dev-Box

Development environments for supported Adhearsion telephony engines using Vagrant
http://adhearsion.github.io/Telephony-Dev-Box
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WebRTC Support #24

Open bklang opened 11 years ago

bklang commented 11 years ago

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).

bklang commented 10 years ago

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

bklang commented 10 years ago

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

bklang commented 9 years ago

See also #50