Multimedia Communication Systems: QoS, RTP, RTCP & RTSP

Explanation

TCP's reliable retransmission mechanism is ill-suited for real-time video calls: retransmitting a lost packet across a 150ms round-trip causes video playback to freeze. Real-time media runs over UDP using RTP (RFC 3550). RTP headers do not guarantee delivery, but provide 16-bit sequence numbers (enabling clients to detect packet loss and reorder packets) and 32-bit timestamps (enabling synchronized playout). Concurrently, RTSP manages the streaming session state over TCP, commanding the media server to stream, pause, or seek without transporting the actual media packets.

Key points

• Jitter Buffers: Client-side playout buffers that temporarily store incoming packets to smooth out variable arrival times before playback.
• Real-Time Transport Protocol (RTP): Runs over UDP; provides payload identification, sequence numbering, and timestamping for synchronization.
• RTP Control Protocol (RTCP): Periodically exchanges transmission statistics (packet loss, jitter, RTT) between sender and receiver for adaptive rate control.
• Real-Time Streaming Protocol (RTSP): Application-level network control protocol acting as an 'internet VCR remote' (PLAY, PAUSE, TEARDOWN, RECORD).

Example

Code examples

/*
RTSP Client-Server Handshake Example:
Client -> Server:
DESCRIBE rtsp://media.example.com/live.mp4 RTSP/1.0
CSeq: 1

Server -> Client:
RTSP/1.0 200 OK
CSeq: 1
Content-Type: application/sdp (Session Description Protocol)

Client -> Server:
SETUP rtsp://media.example.com/live.mp4/trackID=1 RTSP/1.0
CSeq: 2
Transport: RTP/AVP;unicast;client_port=8000-8001

Client -> Server:
PLAY rtsp://media.example.com/live.mp4 RTSP/1.0
CSeq: 3
*/

Exam questions

• [10 Marks] Explain the roles and interactions of RTP, RTCP, and RTSP in multimedia streaming. What fields in the RTP header enable synchronization and jitter mitigation?
• [6 Marks] Define Jitter and describe how a Playout Buffer compensates for packet arrival variance.