RFC 3550 - Real-time Transport Protocol (RTP)
The foundation for real-time multimedia delivery
What is RTP?
The Real-time Transport Protocol (RTP) provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video, or simulation data, over multicast or unicast network services.
Key Features
Payload Identification
Identifies the type of media being transported (audio, video, etc.).
Sequence Numbering
Enables detection of packet loss and out-of-order delivery.
Timestamping
Provides timing information for media synchronization and jitter calculation.
Source Identification
Synchronization source (SSRC) identifiers for multiple streams.
RTP Header Structure
Version (V)
2-bit field identifying the RTP version (currently version 2).
Payload Type (PT)
7-bit field identifying the media format and encoding.
Sequence Number
16-bit field for packet ordering and loss detection.
Timestamp
32-bit field reflecting the sampling instant of the first octet.
SSRC
32-bit synchronization source identifier.
CSRC List
Contributing source identifiers for mixed streams.
RTCP - RTP Control Protocol
RTP is typically used in conjunction with RTCP, which provides:
- Quality Feedback: Statistics on packet loss, jitter, and delay
- Source Description: Canonical name and other participant information
- Session Control: Participant joining and leaving notifications
- Bandwidth Management: Adaptive quality based on network conditions
Common Applications
- VoIP: Voice over IP telephony systems
- Video Conferencing: Real-time video communication
- Live Streaming: Broadcasting audio and video content
- WebRTC: Browser-based real-time communication
- IPTV: Internet Protocol television delivery
- Gaming: Real-time multiplayer game audio/video