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.
