no audio stream upload in webrtc ,bitsSentPerSecond is 0 ,device is available ,need to restart the browser or restart the computer can be restored
Reported by
siheng....@gmail.com,
Apr 12 2017
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Issue description
UserAgent: Mozilla/5.0 (Macintosh; Intel Mac OS X 10_12_2) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/57.0.2987.133 Safari/537.36
Steps to reproduce the problem:
1. call getUserMedia({audio:true,video:true}) to get the stream;
2. RTCPeerconnection.addStream();
3. see in webrtc-internal audio bitsSentPerSecond is 0, audio device is default , and the device is available
What is the expected behavior?
What went wrong?
SDP negotiation is ready, send stream to other client , audio bitsSentPerSecond is 0 ,audio device is available,need to restart the browser or restart the computer can be restored
Did this work before? N/A
Does this work in other browsers? N/A
Chrome version: 57.0.2987.133 Channel: stable
OS Version: OS X 10.12.2
Flash Version:
,
May 20 2017
Closing due to lack of feedback. The description of the bug is not clear enough to be actionable. |
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Comment 1 by rbasuvula@chromium.org
, Apr 13 2017Labels: Needs-Feedback