Based on kurento – asterisk (https://webrtc.ventures/2017/02/kurento-asterisk-powerful-couple/) Study Kurento-Asterisk by testing a call between WebRTC with Kurento App Server to a softphone registered on Asterisk.
Step 1. Javascript SoftPhone using SIP over WebSocket connects to Asterisk and make a phone call with a soft-phone (LinPhone, X-Lite, …)
> Working
- Use sipml5 to test Asterisk configuration. https://www.doubango.org/sipml5/call.htm?svn=170
- Error “SIPml-api.js?svn=250:1 WebSocket connection to ‘wss://192.168.0.18:8089/ws’ failed: WebSocket opening handshake was canceled” then open https://192.168.0.18:8089/ws and clear certification issue
- ctxSip is another WebRTC SIP web phone https://collecttix.github.io/ctxSip/
Step 2. build and run https://github.com/chealwoo/kurento-asterisk.git
> Not working, I think the code is not finished and need to build mediaHandler myself.
- Asterisk Install and Configuration
- https://wiki.asterisk.org/wiki/display/AST/Interactive+Connectivity+Establishment+(ICE)+in+Asterisk
- https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject
Doesn’ work and found the following
http://sipjs.com/guides/server-configuration/asterisk/
Then found some more discussions;
- https://groups.google.com/forum/#!topic/kurento/GqZ_OacEbjM
- https://groups.google.com/forum/#!searchin/kurento/sip%7Csort:date/kurento/viXdmT49M-c/-UP2ewPiBwAJ
I still have the error below
[Mar 26 22:51:48] WARNING[8452]: netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null)
> 0x7f8f5000b870 — Probation passed – setting RTP source address to 192.168.0.12:51898
2017-3-27
More practical post of what I am doing
Establish connection between Asterisk and Kurento https://groups.google.com/forum/#!topic/kurento/0qChzY85Nr0