Technical Manual Signaling & Protocols

HUAWEI MSOFTX3000 Mobile SoftSwitch Center

Table of Contents

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

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Chapter 2 RTP and RTCP

Chapter 2 RTP and RTCP

2.1 Brief Introduction
The IP bearer voice services are transmitted based on UDP, which however does not
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, the IETF defines the Real-time Transport Protocol
(RTP).
RTP provides real-time end-to-end data delivery services, such as interactive audio
and video files.
Such services include

Payload type identification

Sequence numbering

Time stamping

Delivery monitoring.

RTP itself does not provide any mechanism to ensure timely delivery or provide other
quality of service (QoS) 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.

RTP: transports the information featured with real time.

RTP Control Protocol (RTCP): monitors QoS 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, refer to
related RFC documents. RTP does not address resource reservation and does not
guarantee QoS for real-time services. The data transport is augmented by RTCP. The
RTCP allows monitoring of the data delivery in a manner scalable to large multicast
networks, and provides minimal control and identification. RTP/RTCP is independent
from the following transport layer and network layer.

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Chapter 2 RTP and RTCP

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 Figure 1.1.
MRPU
Nb UP
RTP

RTCP
UDP
IP
ETH

IP

Figure 1.1 Applications 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 Table 1.1.
Table 1.1 The meaning of RTP header fields
Field

Length

Meaning

(bit)
Version

The field defines the version of RTP. The version defined

here is two.

Padding(P)

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.

Extension (X)

If the extension bit is set to 1, the RTP header is

followed by exactly one header extension.

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Field

Chapter 2 RTP and RTCP

Length

Meaning

(bit)
CSRC Count

The CSRC count contains the number of CSRC

(CC)
Mark (M)

identifiers that follow the header.

1

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.

Payload

This field identifies the format of the RTP payload.

16

The sequence number is used by the receiver to detect

Type (PT)
Sequence
Number

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.

Timestamp

32

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.
SSRC

32

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 mu
ltiple 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.

CSRC List

0-480

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

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Chapter 2 RTP and RTCP

2.3.2 RTCP Packet Format

RTCP defines several types of RCTP packets to carry a variety of control information
as shown in Table 1.1.
Table 1.1 RTCP packets
Control information
SR (sender report)

Description
Describes transmission and reception statistics from the
gateways that are active senders

RR (receiver report)

Describes reception statistics from the gateways that are

receivers

SDES (source

Describes the sources sending RTCP packets, including

description item)

CNAME

BYE

Indicates end of voice transfer

APP

Application specific functionality extension

Each RTCP packet begins with a fixed part similar to that of RTP data packets; and it
is 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 and a length field in
the fixed part make RTCP packets "stackable". In other words, 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 performs two functions as
follows:

Provides 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.

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 discovered or
a program is restarted, receivers require the CNAME to keep track of each partici
pant.

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Chapter 2 RTP and RTCP

2.3.4 RTCP Transmission Interval

The interval between RTCP packets transmitted is varied randomly over the range
0.51.5 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
random variation of half the minimum RTCP interval in case the application is started
at multiple sites at the same time.

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