WebRTC

WebRTC/rtcweb is an effort to bring a defined API to JavaScript

developers that allows them to venture into the world of real time communications.

Asterisk module

res_http_websocket

TURN support has been added to res_rtp_asterisk to allow clients behind NAT to better communicate with Asterisk.

Browser Support

https://en.wikipedia.org/wiki/WebRTC

設定

load modules

Also ensure that res_http_websocket.so is loaded prior to chan_sip.so if you are not using autoload in modules.conf

boxA*CLI> module show like res_http

builtin webserver

http.conf

[general]
bindaddr=0.0.0.0
enabled=yes
bindport=8088

測試

http://192.168.88.81:8088/

sip.conf

; To allow a peer, user, or friend access using the WebSocket transport

transport=udp,ws,wss

; added to the peer, user, or friend.
; The WebRTC standard has selected AVPF as the audio video profile to use for media streams.

avpf=yes

Webct Client

http://sipml5.org/

apt-get install subversion

svn checkout http://sipml5.googlecode.com/svn/trunk/ sipml5-read-only