Transport-CC (defined in draft-holmer-rmcat-transport-wide-cc-extensions, originally a Google contribution, widely deployed since 2016) is the per-packet feedback mechanism that powers modern WebRTC bandwidth estimation. Where classical RTCP receiver reports give coarse loss/jitter statistics every few seconds, Transport-CC sends a compact RTCP message reporting the precise arrival time of every recently-received RTP packet, every few hundred milliseconds.
The sender uses this stream of timing data to estimate bottleneck bandwidth and queueing delay. Google's GCC (Google Congestion Control) algorithm — the WebRTC default in Chrome and Firefox — uses Transport-CC feedback to adjust send rate continuously. The result is much more responsive bandwidth adaptation than classical RTCP allows, which is essential for sub-second-latency WebRTC where there's no buffer to absorb mistakes.
Transport-CC is asymmetric — the receiver sends rich feedback, the sender uses it to adjust. Both ends need to support it (negotiated via SDP transport-cc extension and the goog-remb / goog-cc lines). Every modern WebRTC stack supports Transport-CC by default; older or specialised stacks may fall back to receiver-side bandwidth estimation, which works but with less precision. Custom WebRTC pipelines for low-latency live streaming often write their own bandwidth estimators that consume Transport-CC feedback directly.
