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02-Chapter 2 RTP and RTCP

02-Chapter 2 RTP and RTCP

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Technical Manual – Signaling & Protocols

Table of Contents

Table of Contents
Chapter 2 RTP and RTCP.................................................................................................... .........1 2.1 Brief Introduction.................................................................................................. .............1 2.2 RTP/RTCP Applications .............................................................................. .....................2 2.3 Packet Format and Meaning ........................................................................... .................2 2.3.1 RTP Header Format .................................................................. ............................2 2.3.2 RTCP Packet Format........................................................................................ ......3 2.3.3 RTCP Functions........................................................................... ..........................4 2.3.4 RTCP Transmission Interval................................................................ ...................4

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Chapter 2 RTP and RTCP
2.1 Brief Introduction
The IP bearer voice services are transmitted based on UDP, which however doesn’t take real-time service transmission into account (media stream synchronization, for example), as it is designed to be dedicated to data stream transmission. So the functionality of UDP needs to be expanded when real-time services are transferred based on it. For that purpose, IETF defines a new protocol RTP (Real-time Transport Protocol). RTP provides real-time end-to-end data delivery services, such as interactive audio and, including payload type identification, sequence numbering, time stamping and delivery monitoring. RTP itself does not provide any mechanism to ensure timely delivery or provide other quality-of-service guarantees, but relies on lower-layer protocols to do so. At present, RTP/RTCP is widely used in IP bearer voice service flow transmission. RTP can also provide end-to-end network transport functions over multicast or unicast network services, which are suitable for applications transmitting various real-time data, such as video and simulation data. RTP involves two closely correlated parts.

Real-time Transport Protocol (RTP): transports the information featured with real time. RTP Control Protocol (RTCP): monitors quality of service and the information about members involving a transfer or a session.

RTP guarantees timely transmission and synchronization of audio and video; and RCTP is used to monitor RTP and QoS. For details about the protocols, please refer to related RFC documents. RTP does not address resource reservation and does not guarantee quality-of-service for real-time services. The data transport is augmented by RTCP to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP/RTCP is designed to be independent from the underlying transport layer and network layer.

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2.2 RTP/RTCP Applications
The IP bearer of voice services is mainly accomplished by RTP. In the UMG8900 device, as for bearer handovers between ATM to IP and between TDM to IP, RTP/RTCP is responsible for IP bearer service processing and adaptation. The functions of RTP/TCP are provided and fulfilled by MRPU of the UMG8900 device. The application of RTP/RTCP in MRPU is shown in 2.2.

MRPU Nb UP RTP UDP IP ETH IP

RTCP

Figure 1.1Applications of RTP/RTCP in the UMG8900 device

RTP/RTCP is the protocol on top of the transport layer. RTP accomplishes Nb UP adaptation and RTCP monitors RTP packets.

2.3 Packet Format and Meaning
2.3.1 RTP Header Format
A RTP header contains many fields as shown in 2.3.1. Table 1.1The meaning of RTP header fields Field Version Padding(P) Length (bit) 2 1 Meaning The field defines the version of RTP. The version defined here is two. If the padding bit is set to 1, the packet contains one or more additional padding octets at the end of the header. The last octet of the padding contains a count of padding octets. Padding may be needed by some encryption algorithms or for carrying several RTP packets in a lower-layer protocol data packet.

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Field Extension (X) CSRC Count (CC) Mark (M)

Length (bit) 1 4 1

Meaning If the extension bit is set to 1, the RTP header is followed by exactly one header extension. The CSRC count contains the number of CSRC identifiers that follow the header. It is set by specific protocols. In IP calls, it is set to 1 in the first RTP data packet transferred after mute, and it is set to 0 in other cases. This field identifies the format of the RTP payload. The sequence number is used by the receiver to detect packet loss and to restore packet sequence. The initial value of the sequence number is random and increments by one for each RTP data packet sent. The timestamp reflects the sampling instant of the first octet in the RTP data packet. The sampling instant must accommodate to synchronization to allow synchronization and jitter calculations. The initial value of the timestamp is random, and increments with the size of packet data. The SSRC field identifies the RTP packet sender. This identifier is chosen randomly, with the intent that no two RTP packet senders within the same gateway will have the same SSRC identifier. Although the probability of mul tiple sources choosing the same identifier is low, all RTP implementations must be prepared to detect and resolve collisions. If a source changes its source transport addre ss, it must also choose a new SSRC identifier to avoid b eing interpreted as a looped source. 0 to 15 items, 32 bits each. The CSRC list identifies CSRC in packets. The number of identifiers is given by the CC field. At most 15 CSRC identifiers are defined and are inserted by mixers, using the SSRC identifiers

Payload Type (PT) Sequence Number

7 16

Timestamp

32

SSRC

32

CSRC List

0-480

2.3.2 RTCP Packet Format
RTCP defines several types of RCTP packets to carry a variety of control information as shown in 2.3.2.

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Table 1.2RTCP packets Control information SR (sender report) RR (receiver report) SDES (source description item) BYE APP Description Describe transmission and reception statistics from the gateways that are active senders Describe reception statistics from the gateways that are receivers Describe the sources sending RTCP packets, including CNAME. Indicates end of voice transfer. Application specific functionality extension.

Each RTCP packet begins with a fixed part similar to that of RTP data packets, followed by structured elements that may be of variable length according to the packet type but always end on a 32-bit boundary. The alignment requirement and a length field in the fixed part make RTCP packets "stackable", that is, multiple RTCP packets form a compound RTCP packet that is sent in a single packet of the lower layer protocol, for example UDP. There is no explicit count of individual RTCP packets in the compound packet since the lower layer protocols are expected to provide an overall length to determine the end of the compound packet.

2.3.3 RTCP Functions
RTCP transmits RTP control packets based on the periodic transmission, using the same distribution mechanism as the data packets. RTCP chiefly performs two functio ns.

The primary function is to provide feedback on the quality of the data distribution. The receiver diagnoses faults on transport lines and controls RTP packet transfer according to the feedback information in RTCP packets. The feedback function is accomplished through sending and receiving reports by RTCP.

RTCP carries a persistent identifier for a RTP source, which is called the canonical name (CNAME). Since the RTP header may change if a conflict is disc overed or a program is restarted, receivers require the CNAME to keep track of e ach participant.

2.3.4 RTCP Transmission Interval
The interval between RTCP packets transmitted is varied randomly over the range

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[0.5, 1.5] times the calculated interval to avoid unintended synchronization of all participants. The first RTCP packet sent after joining a session is also delayed by a ra ndom variation of half the minimum RTCP interval in case the application is started at multiple sites simultaneously.

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