What is RTCP?
RTCP (RTP Control Protocol) is an auxiliary protocol that works alongside RTP (Real-time Transport Protocol). It doesn’t transmit media, but monitors transmission quality by collecting real-time telemetry.
Why is RTCP needed?
Streaming audio and video (VoIP, WebRTC, IPTV) are sensitive to delay and packet loss. RTCP enables:
- Tracking packet loss and jitter
- Measuring RTT and average latency
- Reporting bitrate and stream configuration
How RTCP works
Types of RTCP packets
- Sender Report (SR) — report on transmission (time, bytes sent)
- Receiver Report (RR) — report on reception (loss, delay)
- SDES (Source Description) — source name, role
- BYE — session termination
- APP — application-specific extensions
Practical Usage
| Domain | Protocols | RTCP Role |
|---|---|---|
| VoIP (SIP) | RTP + RTCP | Loss monitoring, MOS dynamics |
| WebRTC | RTP/RTCP/DTLS | Adaptive video transmission |
| Video conferencing | H.323, Zoom, Teams | QoS and FEC statistics |
Guide: How to Monitor RTCP
- Start Wireshark, filter:
<span>udp.port == 5005</span> - Find RTCP RR and SR packet types
- Compare jitter, loss, RTT
Tip: RTCP is transmitted via UDP, usually on a port adjacent to RTP
FAQ
Is RTCP mandatory?
No, but highly recommended. Without it, it’s impossible to monitor real connection quality or adapt codecs.
Is RTCP encrypted?
Yes, when SRTP/DTLS is used. In WebRTC, all RTCP traffic is secured.
What’s the difference from RTP?
RTP carries the media (audio, video), while RTCP provides metadata about its transmission.
Conclusion
RTCP is an essential part of modern streaming communication. It enables not only monitoring but also real-time adaptation of network application behavior. Without it, efficient VoIP and WebRTC operation in unstable networks is impossible.
