Handle retransmission in WebRTC audio jitter buffer |
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Issue descriptionCreate origin trial for handling retransmitted packets in WebRTC audio jitter buffer. Intent to experiment: https://groups.google.com/a/chromium.org/forum/#!topic/blink-dev/A31F88qhQLQ |
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Comment 1 by bugdroid1@chromium.org
, Yesterday (36 hours ago)