webrtc-sip

webrtc jssip prototype

Must add SIP server configuration at index.js before testing.

var configuration = {
    'uri': 'sip:user@domain',
    'password': '', // FILL PASSWORD HERE,
    'ws_servers': 'wss://<ws server ip or fqdn>:<port>/'
};

JsSIP version

The original prototype was based on JsSIP 0.7.x. The branch is jssip0.7.

Since JsSIP 3.x, addstream in the session event has been removed. In order to acquire local and remote streams, we have to add listeners to RTCPeerConnections directly. This differs from JsSIP 0.7 version.

How to start

Open index.html in Chrome.

supported use cases

It's only tested on Chrome on CentOS

1. Outbound/Inbound call to a North American number
2. Outbound/Inbound call to an extension
3. Video call from/to Monster UI WebRTC
4. Conference call(the remote video would the same as the local video feed)

Known bugs

1. If the remote terminates an outbound call, JsSIP will not trigger the right callback. But this seems FS issue.

JsSIP:Transport received WebSocket text message: SIP/2.0 486 Unable to Comply

https://groups.google.com/forum/#!topic/2600hz-dev/krikOmTA61E