Detected major WebRTC bug in Chrome browser related Opus codec

The bug is detected in the latest Chrome 39.0.2171.71 m version. If the Opus WebRTC stream being played by Chrome contains inband FFC(Forward Error Correction) information, the stream will not be played correct (very high jitter, robotic voice, no audio at all, etc). The issue was fixed in Chromium 40.0.2211.0.

We tested the issue with Jitsi client which allow to disable FFC for Opus codec.

Additional information: https://code.google.com/p/chromium/issues/detail?id=423985

Resolution:

The issue affects Chrome-SIP calls.

1. You can resolve it using SIP endpoints without FFC feature for Opus.

or

2. You can remove opus codec from flashphoner settings and use G.711 only.
codecs=alaw,ulaw,speex16,g729,telephone-event

or

3. You can wait for stable Chrome version containing this fix (from Chromium branch).