10.1 REAL-TIME PROTOCOL (RTP)
RTP is the real-time protocol used with voice, audio, video, and several other real-time media-specific applications. RTP carries media payload and realtime parameters for extracting the timing details. RTP parameters are also used in jitter buffer adjustment in VoIP applications. VoIP packets are framed with compressed voice payload, RTP, UDP, and IP header combinations as shown in Fig. 10.1 and Chapter 11. RTP and UDP combinations are used for real-time voice delivery. The VoIP packet consists of an IP header of 20 bytes given in RFC0791 [URL (RFC791)] and a UDP header of 8 bytes given in RFC0768 [Postel (1980)]. Real-time aspects of transport are taken care of by a 12-byte header of RTP RFC3550 [Schulzrinne et al. (2003)]. Voice is sensitive to delays, and RTP helps proper end-to-end delivery of real-time voice traffic. RTP header compression reduces the number of bytes. RTP header compression [Casner and Jacobson (1999)] is not considered in this chapter. The RTP header format is illustrated in Fig. 10.1 [Schulzrinne et al. (2003)].

Figure 10.1. RTP and RTCP packet formats. (a) RTP packet format. (b) RTCP header. (c) RTCP packet format at a high level.
The first 12 octets up to synchronization source (SSRC) are present in every RTP packet, whereas the list of the contributing source (CSRC) identifiers is present only when inserted by a mixer. RTP ...
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