Media Transport for VoIP

Abstract

TCP and UDP are the most commonly used transport layer protocols. TCP is a connection-oriented, reliable, in-order transport protocol. Its features such as retransmission, flow control and congestion control are not suitable for real-time multimedia applications such as VoIP. UDP is a connectionless and unreliable transport protocol. Its simple header, non-retransmission and non-congestion-control features make it suitable for real-time applications. However, as UDP does not have the sequence number in the UDP header, the media stream packet transferred over UDP may experience duplication or arrive not in the right order. This will cause the received media (e.g., voice or video) unrecognisable or unviewable. The Real-time Transport Protocol (RTP) was developed to assist the transfer of real-time media streams on top of the unreliable UDP protocol. It has many fields, such as the sequence number (to detect packet loss), the timestamp (to know the location of media packet) and the payload type (to know voice or video codec used). The associated RTP Control Protocol (RTCP) was also developed to assist media control and QoS/QoE management for VoIP applications. This chapter presents the key concepts of RTP and RTCP, together with detailed header analysis based on real trace data using Wireshark. The compressed RTP (cRTP) and bandwidth efficiency issues are also discussed together with illustrative worked examples for VoIP bandwidth calculation.