An RTCP Sender Report (SR) is the small control packet that makes lip-sync possible in RTP-based systems like WebRTC. Each media stream — audio and video — runs on its own RTP timestamp clock, and those clocks can't be compared directly. The sender report periodically carries a pairing: this RTP timestamp corresponds to this absolute NTP wall-clock time. The receiver collects an SR from each stream, converts both RTP timelines onto the common NTP reference, and can then schedule audio and video to be presented together. Sender reports also carry transmission statistics, but their synchronization role is the one that decides whether a video call's voice matches the lips.
The book · Volume 5 of 7
Audio for Video: A Complete Guide to Sound in Video Products: From Jitter Buffers and Lip Sync to Atmos, Quality Metrics, and Neural Codecs
The RTCP Sender Report is the single packet that links RTP time to wall-clock time, and without it a receiver cannot line up audio and video. This volume explains Sender Reports in detail with RTP timestamps, NTP, clock domains and lip sync, so you can verify that every stream in a call is synced correctly. Written by Nikolay Sapunov, CEO at Fora Soft.
Lip sync failing in WebRTC or RTP calls because RTCP Sender Reports are wrong or missing?
Share packet captures and we will check your RTCP Sender Report timing and NTP mapping and give you a sync fix plan. 250+ video products shipped since 2005.
