Webrtc

Starting from version 3.9, liblinphone is inter-workable with both Google and Firefox WebRTC engines. This has been achieved by adding the following new capabilities to liblinphone/mediastremer2/ortp:

Compilation

Lliblinphone is compiled with DTLS support enabled by default. Since March  2015 this feature is included in the source code distribution.

Support can be checked at runtime by using the function:

Configuration

  • Enable both OPUS and VP8 codecs

Limitations

Liblinphone does not support bundled audio/video. This feature must be disabled in webrtc configuration.

Optimizations

With some Webrtc/sip stack integration it might be useful to increase ICE candidates gathering time by settings linphonerc parameters:

[sip]
delayed_timeout=60

rtcp-mux is also required by some implemention

[rtp]

rtcp_mux=1

Debugging

Enabling Chrome debug traces

Chrome --enable-logging --v=4
Created by SandrineAvakian on 2017/01/05 10:47