The IP is a relatively low-level protocol. It was originally developed for delivery of packets (or datagrams) between host computers across the ARPAnet (Internet) packet network. For an IP telephony application, datagrams are transmitted between desktop voice terminals. IP is a connectionless protocol that does not establish a virtual connection through a network before commencing transmission. Establishing a communications path between endpoints is the responsibility of higherlevel protocols.
IP makes no guarantees concerning reliability, flow control, error detection, or error correction. As a result, datagrams could arrive at the destination computer out of sequence, with errors, or not even arrive at all. This is known as jitter. IP does succeed in making the network transparent to the upper layers involved in voice transmission through an IPbased network.
VoIP transmission, by definition, uses IP, although it is not well suited for voice transmission. Real-time applications such as voice and video require guaranteed connection with consistent delay characteristics. Higher-layer protocols address these issues. There are two available protocols at the transport layer when transmitting information through an IP network. These are Transmission Control Protocol (TCP) and User Datagram Protocol (UDP). Both protocols enable the transmission of information between the correct processes (or applications) on network endpoints. These processes are associated with unique port numbers.
TCP is a connection-oriented protocol. It establishes a communications path before transmitting data. It handles sequencing and error detection, thus ensuring that a reliable stream of data is received by the destination application. TCP can address real-time voice applications to a certain extent but would require higher-layer functions. Voice applications require that information is received in the correct sequence, reliably, and with predictable delay characteristics. With this in mind, the ITU-T decided that the alternative protocol, UDP, should be used. UDP is also a connectionless protocol. UDP routes data to its correct destination port but does not attempt to perform any sequencing or ensure data reliability.
To provide feedback on the quality of the transmission link, the RTP/RFTP protocols, developed by the IETF, are used. Real-Time Transport Protocol (RTP) transports the digitized samples of real-time information, and Real-Time Control Protocol (RTCP) provides the mechanism for quality feedback. RTP and RTCP do not reduce the overall delay of the real-time information. Nor do they make any guarantees concerning QoS.
When an IP voice terminal transmits datagrams across the LAN/WAN, the IP, UDP, and RTP headers are followed by the data payload of the RTP header. The data payload is comprised of digitized voice samples. The length of these samples can vary, but for voice samples representing 20 ms are considered the maximum duration for the payload. The number of transmitted datagrams varies indirectly with the sampling rate—the longer the sampling period, the fewer the number of packets transmitted per second. The selection of the payload duration is a compromise between bandwidth requirements and quality. Smaller payloads demand higher bandwidth per channel band because the header length remains at 40 octets. However, if payloads are increased, the overall delay of the system will increase, and the system will be more susceptible to the loss of individual packets by the network.
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