How Do You Synchronize Audio and Video in Real-Time Streams?

Audio and video desynchronization in real-time streaming systems is a complex issue caused by three main factors: clock differences during capture, asymmetric encoding pipelines, and network jitter. Audio and video are captured on separate hardware with independent clocks, which can drift over time, leading to synchronization issues. Encoding asymmetry arises because audio and video codecs operate on different timescales, with audio having a consistent packet size and frequency, while video encoding varies greatly depending on frame content. Once on the network, audio and video packets contend for bandwidth, with video often experiencing more variable delays, especially during keyframe intervals. WebRTC addresses these synchronization challenges through RTP timestamps and RTCP Sender Reports, which synchronize audio and video streams by mapping their RTP timestamps to a common NTP wall-clock reference. Jitter buffers in WebRTC further help manage network arrival variations but can introduce sync issues if audio and video buffers add different delays. Selective Forwarding Units (SFUs) in deployments further complicate synchronization by generating their own RTCP Sender Reports, which may introduce asymmetries not present in direct peer-to-peer connections. Observing metrics like jitter, jitter buffer delay, and packet loss through browser APIs such as getStats() is crucial for diagnosing and addressing AV sync issues in WebRTC implementations.