webrtc - How does audio and video in a webrtc peerconnection stay in sync?

Question

How does audio and video in a webrtc peerconnection stay in sync? I am using an API which publishes audio and video (I assume as one peer connection) to a media server. The audio can occasionally go out of sync up to 200ms. I am attributing this to the possibility that the audio and video are separate streams and this accounts for the why the sync can be out.

标签： webrtcrtcpeerconnection

Answer 1

RTP/RTCP（WebRTC 使用）传统上使用RTCP Sender Report。这允许每个 SSRC 流在 NTP 时间戳上同步。浏览器今天确实使用它们，所以应该可以。

你在做任何协议桥接或任何可能只是 RTP 的事情吗？您使用的是什么媒体服务器？