อัตราบิตแบบปรับตามสถานการณ์และการควบคุมความคับคั่ง

Real-time media runs over networks that fluctuate constantly: Wi-Fi dips, cellular handoffs, shared bandwidth. A fixed bitrate will either waste capacity or overwhelm a weak link.

Adaptive bitrate lets the sender match its output to current conditions.

WebRTC infers congestion from feedback:

• Packet loss reported via RTCP.
• Increasing delay (jitter and one-way delay growth).
• Receiver estimates of available bandwidth.

The core algorithm in WebRTC is GCC (Google Congestion Control). It combines a loss-based controller and a delay-based controller to estimate the safe sending rate.

The delay-based part watches how packet arrival times drift. If packets start queuing up, that growing delay signals congestion before loss even happens, so the sender can back off early.

The loss-based controller reacts to RTCP loss reports. Above a threshold it lowers the rate; with little loss it slowly probes upward to reclaim bandwidth.

When bandwidth drops, the encoder must shed bits. It can lower resolution (softer image) or frame rate (choppier motion). WebRTC chooses based on content: video calls favor smoothness, screen shares favor sharpness.

With simulcast, a sender encodes the same video at several qualities at once. The SFU forwards the layer each receiver can handle, so one weak viewer does not degrade everyone.

You can observe the estimate via getStats(), watching available outgoing bitrate and current send rate to drive UI warnings.

You can bound the encoder so it never exceeds a ceiling, useful for cost control or guaranteeing other traffic gets bandwidth.

To survive loss without dropping quality, WebRTC uses FEC (forward error correction) and selective retransmission (NACK/RTX). These add overhead but protect against lossy links.

• Conferencing: prioritize low latency and smoothness.
• Broadcast: allow a larger buffer for stability.
• Screen share: protect resolution over frame rate.

WebRTC adapts bitrate using GCC's delay- and loss-based estimates, trades resolution for frame rate as needed, and uses simulcast plus FEC/retransmission to keep calls smooth across unstable networks.