VoIP Voice and Fax Signal Processing

10

PACKETIZATION—RTP, RTCP, AND JITTER BUFFER

In VoIP, voice samples from telephone interfaces are compressed using compression codecs such as G.711, G.729A, G.723.1, and G.722, and are framed as payload. Voice payload size varies with the compression codec, compression rate options, and payload duration. For G.729A, 10 ms is the frame, and for G.723.1,30 ms is the basic frame. Voice payload may use a group of compressed frames up to 80 ms. The compressed payloads are framed as Real-Time Protocol (RTP)/User Protocol Datagram (UDP)/Internet Protocol (IP) packets for sending on the IP network. At the destination, payloads from RTP/UDP/IP are passed through the jitter buffer before decompressing in the decoder. RTP Control Protocol (RTCP), RTCP-Extended Report (XR) packets are used to convey end-to-end voice packet transmission parameters and statistics. In actual implementation, voice payload, RTP, jitter buffer, RTCP, voice quality monitoring, quality of service (QoS) mechanisms, and bandwidth management parameters work in coordination for ensuring better end-to-end packet delivery. This chapter is presented for RTP, RTCP, RTCP-XR, packet impediments, and jitter buffer. More details on voice packet headers of UDP/IP and network interface headers for Ethernet and digital subscriber line (DSL) are presented in Chapter 11.