Request for Comments: 3550 Columbia University

 

Network Working Group H. Schulzrinne

Request for Comments: 3550 Columbia University

Obsoletes: 1889 S. Casner

Category: Standards Track Packet Design

R. Frederick

Blue Coat Systems Inc.

V. Jacobson

Packet Design

July 2003

RTP: A Transport Protocol for Real-Time Applications

Status of this Memo

This document specifies an Internet standards track protocol for the

Internet community, and requests discussion and suggestions for

improvements. Please refer to the current edition of the "Internet

Official Protocol Standards" (STD 1) for the standardization state

and status of this protocol. Distribution of this memo is unlimited.

Network Communication Protocol Map. To order: http://www.javvin.com/map.html

Easy to use sniffing tool: http://www.javvin.com/packet.html

Copyright Notice Copyright (C) The Internet Society (2003). All Rights Reserved.

Abstract

This memorandum describes RTP, the real-time transport protocol. RTP

provides end-to-end network transport functions suitable for

applications transmitting real-time data, such as audio, video or

simulation data, over multicast or unicast network services. RTP

does not address resource reservation and does not guarantee

quality-of-service for real-time services. The data transport is

augmented by a control protocol (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 and

RTCP are designed to be independent of the underlying transport and

network layers. The protocol supports the use of RTP-level

translators and mixers.

Most of the text in this memorandum is identical to RFC 1889 which it

obsoletes. There are no changes in the packet formats on the wire,

only changes to the rules and algorithms governing how the protocol

is used. The biggest change is an enhancement to the scalable timer

algorithm for calculating when to send RTCP packets in order to

minimize transmission in excess of the intended rate when many

participants join a session simultaneously.

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Table of Contents

1. Introduction ................................................ 4

1.1 Terminology ............................................ 5

2. RTP Use Scenarios ........................................... 5

2.1 Simple Multicast Audio Conference ...................... 6

2.2 Audio and Video Conference ............................. 7

2.3 Mixers and Translators ................................. 7

2.4 Layered Encodings ...................................... 8

3. Definitions ................................................. 8

4. Byte Order, Alignment, and Time Format ...................... 12

5. RTP Data Transfer Protocol .................................. 13

5.1 RTP Fixed Header Fields ................................ 13

5.2 Multiplexing RTP Sessions .............................. 16

5.3 Profile-Specific Modifications to the RTP Header ....... 18

5.3.1 RTP Header Extension ............................ 18

6. RTP Control Protocol -- RTCP ................................ 19

6.1 RTCP Packet Format ..................................... 21

6.2 RTCP Transmission Interval ............................. 24

6.2.1 Maintaining the Number of Session Members ....... 28

6.3 RTCP Packet Send and Receive Rules ..................... 28

6.3.1 Computing the RTCP Transmission Interval ........ 29

6.3.2 Initialization .................................. 30

6.3.3 Receiving an RTP or Non-BYE RTCP Packet ......... 31

6.3.4 Receiving an RTCP BYE Packet .................... 31

6.3.5 Timing Out an SSRC .............................. 32

6.3.6 Expiration of Transmission Timer ................ 32

6.3.7 Transmitting a BYE Packet ....................... 33

6.3.8 Updating we_sent ................................ 34

6.3.9 Allocation of Source Description Bandwidth ...... 34

6.4 Sender and Receiver Reports ............................ 35

6.4.1 SR: Sender Report RTCP Packet ................... 36

6.4.2 RR: Receiver Report RTCP Packet ................. 42

6.4.3 Extending the Sender and Receiver Reports ....... 42

6.4.4 Analyzing Sender and Receiver Reports ........... 43

6.5 SDES: Source Description RTCP Packet ................... 45

6.5.1 CNAME: Canonical End-Point Identifier SDES Item . 46

6.5.2 NAME: User Name SDES Item ....................... 48

6.5.3 EMAIL: Electronic Mail Address SDES Item ........ 48

6.5.4 PHONE: Phone Number SDES Item ................... 49

6.5.5 LOC: Geographic User Location SDES Item ......... 49

6.5.6 TOOL: Application or Tool Name SDES Item ........ 49

6.5.7 NOTE: Notice/Status SDES Item ................... 50

6.5.8 PRIV: Private Extensions SDES Item .............. 50

6.6 BYE: Goodbye RTCP Packet ............................... 51

6.7 APP: Application-Defined RTCP Packet ................... 52

7. RTP Translators and Mixers .................................. 53

7.1 General Description .................................... 53

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7.2 RTCP Processing in Translators ......................... 55

7.3 RTCP Processing in Mixers .............................. 57

7.4 Cascaded Mixers ........................................ 58

8. SSRC Identifier Allocation and Use .......................... 59

8.1 Probability of Collision ............................... 59

8.2 Collision Resolution and Loop Detection ................ 60

8.3 Use with Layered Encodings ............................. 64

9. Security .................................................... 65

9.1 Confidentiality ........................................ 65

9.2 Authentication and Message Integrity ................... 67

10. Congestion Control .......................................... 67

11. RTP over Network and Transport Protocols .................... 68

12. Summary of Protocol Constants ............................... 69

12.1 RTCP Packet Types ...................................... 70

12.2 SDES Types ............................................. 70

13. RTP Profiles and Payload Format Specifications .............. 71

14. Security Considerations ..................................... 73

15. IANA Considerations ......................................... 73

16. Intellectual Property Rights Statement ...................... 74

17. Acknowledgments ............................................. 74

Appendix A. Algorithms ........................................ 75

Appendix A.1 RTP Data Header Validity Checks ................... 78

Appendix A.2 RTCP Header Validity Checks ....................... 82

Appendix A.3 Determining Number of Packets Expected and Lost ... 83

Appendix A.4 Generating RTCP SDES Packets ...................... 84

Appendix A.5 Parsing RTCP SDES Packets ......................... 85

Appendix A.6 Generating a Random 32-bit Identifier ............. 85

Appendix A.7 Computing the RTCP Transmission Interval .......... 87

Appendix A.8 Estimating the Interarrival Jitter ................ 94

Appendix B. Changes from RFC 1889 ............................. 95

References ...................................................... 100

Normative References ............................................ 100

Informative References .......................................... 100

Authors' Addresses .............................................. 103

Full Copyright Statement ........................................ 104

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1. Introduction

This memorandum specifies the real-time transport protocol (RTP),

which provides end-to-end delivery services for data with real-time

characteristics, such as interactive audio and video. Those services

include payload type identification, sequence numbering, timestamping

and delivery monitoring. Applications typically run RTP on top of

UDP to make use of its multiplexing and checksum services; both

protocols contribute parts of the transport protocol functionality.

However, RTP may be used with other suitable underlying network or

transport protocols (see Section 11). RTP supports data transfer to

multiple destinations using multicast distribution if provided by the

underlying network.

Note that RTP itself does not provide any mechanism to ensure timely

delivery or provide other quality-of-service guarantees, but relies

on lower-layer services to do so. It does not guarantee delivery or

prevent out-of-order delivery, nor does it assume that the underlying

network is reliable and delivers packets in sequence. The sequence

numbers included in RTP allow the receiver to reconstruct the

sender's packet sequence, but sequence numbers might also be used to

determine the proper location of a packet, for example in video

decoding, without necessarily decoding packets in sequence.

While RTP is primarily designed to satisfy the needs of multiparticipant

multimedia conferences, it is not limited to that

particular application. Storage of continuous data, interactive

distributed simulation, active badge, and control and measurement

applications may also find RTP applicable.

This document defines RTP, consisting of two closely-linked parts:

o the real-time transport protocol (RTP), to carry data that has

real-time properties.

o the RTP control protocol (RTCP), to monitor the quality of service

and to convey information about the participants in an on-going

session. The latter aspect of RTCP may be sufficient for "loosely

controlled" sessions, i.e., where there is no explicit membership

control and set-up, but it is not necessarily intended to support

all of an application's control communication requirements. This

functionality may be fully or partially subsumed by a separate

session control protocol, which is beyond the scope of this

document.

RTP represents a new style of protocol following the principles of

application level framing and integrated layer processing proposed by

Clark and Tennenhouse [10]. That is, RTP is intended to be malleable

Schulzrinne, et al. Standards Track [Page 4]

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to provide the information required by a particular application and

will often be integrated into the application processing rather than

being implemented as a separate layer. RTP is a protocol framework

that is deliberately not complete. This document specifies those

functions expected to be common across all the applications for which

RTP would be appropriate. Unlike conventional protocols in which

additional functions might be accommodated by making the protocol

more general or by adding an option mechanism that would require

parsing, RTP is intended to be tailored through modifications and/or

additions to the headers as needed. Examples are given in Sections

5.3 and 6.4.3.

Therefore, in addition to this document, a complete specification of

RTP for a particular application will require one or more companion

documents (see Section 13):

o a profile specification document, which defines a set of payload

type codes and their mapping to payload formats (e.g., media

encodings). A profile may also define extensions or modifications

to RTP that are specific to a particular class of applications.

Typically an application will operate under only one profile. A

profile for audio and video data may be found in the companion RFC

3551 [1].

o payload format specification documents, which define how a

particular payload, such as an audio or video encoding, is to be

carried in RTP.

A discussion of real-time services and algorithms for their

implementation as well as background discussion on some of the RTP

design decisions can be found in [11].

1.1 Terminology

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",

"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this

document are to be interpreted as described in BCP 14, RFC 2119 [2]

and indicate requirement levels for compliant RTP implementations.

2. RTP Use Scenarios

The following sections describe some aspects of the use of RTP. The

examples were chosen to illustrate the basic operation of

applications using RTP, not to limit what RTP may be used for. In

these examples, RTP is carried on top of IP and UDP, and follows the

conventions established by the profile for audio and video specified

in the companion RFC 3551.

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2.1 Simple Multicast Audio Conference

A working group of the IETF meets to discuss the latest protocol

document, using the IP multicast services of the Internet for voice

communications. Through some allocation mechanism the working group

chair obtains a multicast group address and pair of ports. One port

is used for audio data, and the other is used for control (RTCP)

packets. This address and port information is distributed to the

intended participants. If privacy is desired, the data and control

packets may be encrypted as specified in Section 9.1, in which case

an encryption key must also be generated and distributed. The exact

details of these allocation and distribution mechanisms are beyond

the scope of RTP.

The audio conferencing application used by each conference

participant sends audio data in small chunks of, say, 20 ms duration.

Each chunk of audio data is preceded by an RTP header; RTP header and

data are in turn contained in a UDP packet. The RTP header indicates

what type of audio encoding (such as PCM, ADPCM or LPC) is contained

in each packet so that senders can change the encoding during a

conference, for example, to accommodate a new participant that is

connected through a low-bandwidth link or react to indications of

network congestion.

The Internet, like other packet networks, occasionally loses and

reorders packets and delays them by variable amounts of time. To

cope with these impairments, the RTP header contains timing

information and a sequence number that allow the receivers to

reconstruct the timing produced by the source, so that in this

example, chunks of audio are contiguously played out the speaker

every 20 ms. This timing reconstruction is performed separately for

each source of RTP packets in the conference. The sequence number

can also be used by the receiver to estimate how many packets are

being lost.

Since members of the working group join and leave during the

conference, it is useful to know who is participating at any moment

and how well they are receiving the audio data. For that purpose,

each instance of the audio application in the conference periodically

multicasts a reception report plus the name of its user on the RTCP

(control) port. The reception report indicates how well the current

speaker is being received and may be used to control adaptive

encodings. In addition to the user name, other identifying

information may also be included subject to control bandwidth limits.

A site sends the RTCP BYE packet (Section 6.6) when it leaves the

conference.

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2.2 Audio and Video Conference

If both audio and video media are used in a conference, they are

transmitted as separate RTP sessions. That is, separate RTP and RTCP

packets are transmitted for each medium using two different UDP port

pairs and/or multicast addresses. There is no direct coupling at the

RTP level between the audio and video sessions, except that a user

participating in both sessions should use the same distinguished

(canonical) name in the RTCP packets for both so that the sessions

can be associated.

One motivation for this separation is to allow some participants in

the conference to receive only one medium if they choose. Further

explanation is given in Section 5.2. Despite the separation,

synchronized playback of a source's audio and video can be achieved

using timing information carried in the RTCP packets for both

sessions.

2.3 Mixers and Translators

So far, we have assumed that all sites want to receive media data in

the same format. However, this may not always be appropriate.

Consider the case where participants in one area are connected

through a low-speed link to the majority of the conference

participants who enjoy high-speed network access. Instead of forcing

everyone to use a lower-bandwidth, reduced-quality audio encoding, an

RTP-level relay called a mixer may be placed near the low-bandwidth

area. This mixer resynchronizes incoming audio packets to

reconstruct the constant 20 ms spacing generated by the sender, mixes

these reconstructed audio streams into a single stream, translates

the audio encoding to a lower-bandwidth one and forwards the lowerbandwidth

packet stream across the low-speed link. These packets

might be unicast to a single recipient or multicast on a different

address to multiple recipients. The RTP header includes a means for

mixers to identify the sources that contributed to a mixed packet so

that correct talker indication can be provided at the receivers.

Some of the intended participants in the audio conference may be

connected with high bandwidth links but might not be directly

reachable via IP multicast. For example, they might be behind an

application-level firewall that will not let any IP packets pass.

For these sites, mixing may not be necessary, in which case another

type of RTP-level relay called a translator may be used. Two

translators are installed, one on either side of the firewall, with

the outside one funneling all multicast packets received through a

secure connection to the translator inside the firewall. The

translator inside the firewall sends them again as multicast packets

to a multicast group restricted to the site's internal network.

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Mixers and translators may be designed for a variety of purposes. An

example is a video mixer that scales the images of individual people

in separate video streams and composites them into one video stream

to simulate a group scene. Other examples of translation include the

connection of a group of hosts speaking only IP/UDP to a group of

hosts that understand only ST-II, or the packet-by-packet encoding

translation of video streams from individual sources without

resynchronization or mixing. Details of the operation of mixers and

translators are given in Section 7.

2.4 Layered Encodings

Multimedia applications should be able to adjust the transmission

rate to match the capacity of the receiver or to adapt to network

congestion. Many implementations place the responsibility of rateadaptivity

at the source. This does not work well with multicast

transmission because of the conflicting bandwidth requirements of

heterogeneous receivers. The result is often a least-common

denominator scenario, where the smallest pipe in the network mesh

dictates the quality and fidelity of the overall live multimedia

"broadcast".

Instead, responsibility for rate-adaptation can be placed at the

receivers by combining a layered encoding with a layered transmission

system. In the context of RTP over IP multicast, the source can

stripe the progressive layers of a hierarchically represented signal

across multiple RTP sessions each carried on its own multicast group.

Receivers can then adapt to network heterogeneity and control their

reception bandwidth by joining only the appropriate subset of the

multicast groups.

Details of the use of RTP with layered encodings are given in

Sections 6.3.9, 8.3 and 11.

3. Definitions

RTP payload: The data transported by RTP in a packet, for

example audio samples or compressed video data. The payload

format and interpretation are beyond the scope of this document.

RTP packet: A data packet consisting of the fixed RTP header, a

possibly empty list of contributing sources (see below), and the

payload data. Some underlying protocols may require an

encapsulation of the RTP packet to be defined. Typically one

packet of the underlying protocol contains a single RTP packet,

but several RTP packets MAY be contained if permitted by the

encapsulation method (see Section 11).

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RTCP packet: A control packet consisting of a fixed header part

similar to that of RTP data packets, followed by structured

elements that vary depending upon the RTCP packet type. The

formats are defined in Section 6. Typically, multiple RTCP

packets are sent together as a compound RTCP packet in a single

packet of the underlying protocol; this is enabled by the length

field in the fixed header of each RTCP packet.

Port: The "abstraction that transport protocols use to

distinguish among multiple destinations within a given host

computer. TCP/IP protocols identify ports using small positive

integers." [12] The transport selectors (TSEL) used by the OSI

transport layer are equivalent to ports. RTP depends upon the

lower-layer protocol to provide some mechanism such as ports to

multiplex the RTP and RTCP packets of a session.

Transport address: The combination of a network address and port

that identifies a transport-level endpoint, for example an IP

address and a UDP port. Packets are transmitted from a source

transport address to a destination transport address.

RTP media type: An RTP media type is the collection of payload

types which can be carried within a single RTP session. The RTP

Profile assigns RTP media types to RTP payload types.

Multimedia session: A set of concurrent RTP sessions among a

common group of participants. For example, a videoconference

(which is a multimedia session) may contain an audio RTP session

and a video RTP session.

RTP session: An association among a set of participants

communicating with RTP. A participant may be involved in multiple

RTP sessions at the same time. In a multimedia session, each

medium is typically carried in a separate RTP session with its own

RTCP packets unless the the encoding itself multiplexes multiple

media into a single data stream. A participant distinguishes

multiple RTP sessions by reception of different sessions using

different pairs of destination transport addresses, where a pair

of transport addresses comprises one network address plus a pair

of ports for RTP and RTCP. All participants in an RTP session may

share a common destination transport address pair, as in the case

of IP multicast, or the pairs may be different for each

participant, as in the case of individual unicast network

addresses and port pairs. In the unicast case, a participant may

receive from all other participants in the session using the same

pair of ports, or may use a distinct pair of ports for each.

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The distinguishing feature of an RTP session is that each

maintains a full, separate space of SSRC identifiers (defined

next). The set of participants included in one RTP session

consists of those that can receive an SSRC identifier transmitted

by any one of the participants either in RTP as the SSRC or a CSRC

(also defined below) or in RTCP. For example, consider a threeparty

conference implemented using unicast UDP with each

participant receiving from the other two on separate port pairs.

If each participant sends RTCP feedback about data received from

one other participant only back to that participant, then the

conference is composed of three separate point-to-point RTP

sessions. If each participant provides RTCP feedback about its

reception of one other participant to both of the other

participants, then the conference is composed of one multi-party

RTP session. The latter case simulates the behavior that would

occur with IP multicast communication among the three

participants.

The RTP framework allows the variations defined here, but a

particular control protocol or application design will usually

impose constraints on these variations.

Synchronization source (SSRC): The source of a stream of RTP

packets, identified by a 32-bit numeric SSRC identifier carried in

the RTP header so as not to be dependent upon the network address.

All packets from a synchronization source form part of the same

timing and sequence number space, so a receiver groups packets by

synchronization source for playback. Examples of synchronization

sources include the sender of a stream of packets derived from a

signal source such as a microphone or a camera, or an RTP mixer

(see below). A synchronization source may change its data format,

e.g., audio encoding, over time. The SSRC identifier is a

randomly chosen value meant to be globally unique within a

particular RTP session (see Section 8). A participant need not

use the same SSRC identifier for all the RTP sessions in a

multimedia session; the binding of the SSRC identifiers is

provided through RTCP (see Section 6.5.1). If a participant

generates multiple streams in one RTP session, for example from

separate video cameras, each MUST be identified as a different

SSRC.

Contributing source (CSRC): A source of a stream of RTP packets

that has contributed to the combined stream produced by an RTP

mixer (see below). The mixer inserts a list of the SSRC

identifiers of the sources that contributed to the generation of a

particular packet into the RTP header of that packet. This list

is called the CSRC list. An example application is audio

conferencing where a mixer indicates all the talkers whose speech

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was combined to produce the outgoing packet, allowing the receiver

to indicate the current talker, even though all the audio packets

contain the same SSRC identifier (that of the mixer).

End system: An application that generates the content to be sent

in RTP packets and/or consumes the content of received RTP

packets. An end system can act as one or more synchronization

sources in a particular RTP session, but typically only one.

Mixer: An intermediate system that receives RTP packets from one

or more sources, possibly changes the data format, combines the

packets in some manner and then forwards a new RTP packet. Since

the timing among multiple input sources will not generally be

synchronized, the mixer will make timing adjustments among the

streams and generate its own timing for the combined stream.

Thus, all data packets originating from a mixer will be identified

as having the mixer as their synchronization source.

Translator: An intermediate system that forwards RTP packets

with their synchronization source identifier intact. Examples of

translators include devices that convert encodings without mixing,

replicators from multicast to unicast, and application-level

filters in firewalls.

Monitor: An application that receives RTCP packets sent by

participants in an RTP session, in particular the reception

reports, and estimates the current quality of service for

distribution monitoring, fault diagnosis and long-term statistics.

The monitor function is likely to be built into the application(s)

participating in the session, but may also be a separate

application that does not otherwise participate and does not send

or receive the RTP data packets (since they are on a separate

port). These are called third-party monitors. It is also

acceptable for a third-party monitor to receive the RTP data

packets but not send RTCP packets or otherwise be counted in the

session.

Non-RTP means: Protocols and mechanisms that may be needed in

addition to RTP to provide a usable service. In particular, for

multimedia conferences, a control protocol may distribute

multicast addresses and keys for encryption, negotiate the

encryption algorithm to be used, and define dynamic mappings

between RTP payload type values and the payload formats they

represent for formats that do not have a predefined payload type

value. Examples of such protocols include the Session Initiation

Protocol (SIP) (RFC 3261 [13]), ITU Recommendation H.323 [14] and

applications using SDP (RFC 2327 [15]), such as RTSP (RFC 2326

[16]). For simple

Schulzrinne, et al. Standards Track [Page 11]

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applications, electronic mail or a conference database may also be

used. The specification of such protocols and mechanisms is

outside the scope of this document.

4. Byte Order, Alignment, and Time Format

All integer fields are carried in network byte order, that is, most

significant byte (octet) first. This byte order is commonly known as

big-endian. The transmission order is described in detail in [3].

Unless otherwise noted, numeric constants are in decimal (base 10).

All header data is aligned to its natural length, i.e., 16-bit fields

are aligned on even offsets, 32-bit fields are aligned at offsets

divisible by four, etc. Octets designated as padding have the value

zero.

Wallclock time (absolute date and time) is represented using the

timestamp format of the Network Time Protocol (NTP), which is in

seconds relative to 0h UTC on 1 January 1900 [4]. The full

resolution NTP timestamp is a 64-bit unsigned fixed-point number with

the integer part in the first 32 bits and the fractional part in the

last 32 bits. In some fields where a more compact representation is

appropriate, only the middle 32 bits are used; that is, the low 16

bits of the integer part and the high 16 bits of the fractional part.

The high 16 bits of the integer part must be determined

independently.

An implementation is not required to run the Network Time Protocol in

order to use RTP. Other time sources, or none at all, may be used

(see the description of the NTP timestamp field in Section 6.4.1).

However, running NTP may be useful for synchronizing streams

transmitted from separate hosts.

The NTP timestamp will wrap around to zero some time in the year

2036, but for RTP purposes, only differences between pairs of NTP

timestamps are used. So long as the pairs of timestamps can be

assumed to be within 68 years of each other, using modular arithmetic

for subtractions and comparisons makes the wraparound irrelevant.

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5. RTP Data Transfer Protocol

5.1 RTP Fixed Header Fields

The RTP header has the following format:

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

|V=2|P|X| CC |M| PT | sequence number |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| timestamp |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| synchronization source (SSRC) identifier |

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

| contributing source (CSRC) identifiers |

| .... |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The first twelve octets are present in every RTP packet, while the

list of CSRC identifiers is present only when inserted by a mixer.

The fields have the following meaning:

version (V): 2 bits

This field identifies the version of RTP. The version defined by

this specification is two (2). (The value 1 is used by the first

draft version of RTP and the value 0 is used by the protocol

initially implemented in the "vat" audio tool.)

padding (P): 1 bit

If the padding bit is set, the packet contains one or more

additional padding octets at the end which are not part of the

payload. The last octet of the padding contains a count of how

many padding octets should be ignored, including itself. Padding

may be needed by some encryption algorithms with fixed block sizes

or for carrying several RTP packets in a lower-layer protocol data

unit.

extension (X): 1 bit

If the extension bit is set, the fixed header MUST be followed by

exactly one header extension, with a format defined in Section

5.3.1.

CSRC count (CC): 4 bits

The CSRC count contains the number of CSRC identifiers that follow

the fixed header.

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marker (M): 1 bit

The interpretation of the marker is defined by a profile. It is

intended to allow significant events such as frame boundaries to

be marked in the packet stream. A profile MAY define additional

marker bits or specify that there is no marker bit by changing the

number of bits in the payload type field (see Section 5.3).

payload type (PT): 7 bits

This field identifies the format of the RTP payload and determines

its interpretation by the application. A profile MAY specify a

default static mapping of payload type codes to payload formats.

Additional payload type codes MAY be defined dynamically through

non-RTP means (see Section 3). A set of default mappings for

audio and video is specified in the companion RFC 3551 [1]. An

RTP source MAY change the payload type during a session, but this

field SHOULD NOT be used for multiplexing separate media streams

(see Section 5.2).

A receiver MUST ignore packets with payload types that it does not

understand.

sequence number: 16 bits

The sequence number increments by one for each RTP data packet

sent, and may be used by the receiver to detect packet loss and to

restore packet sequence. The initial value of the sequence number

SHOULD be random (unpredictable) to make known-plaintext attacks

on encryption more difficult, even if the source itself does not

encrypt according to the method in Section 9.1, because the

packets may flow through a translator that does. Techniques for

choosing unpredictable numbers are discussed in [17].

timestamp: 32 bits

The timestamp reflects the sampling instant of the first octet in

the RTP data packet. The sampling instant MUST be derived from a

clock that increments monotonically and linearly in time to allow

synchronization and jitter calculations (see Section 6.4.1). The

resolution of the clock MUST be sufficient for the desired

synchronization accuracy and for measuring packet arrival jitter

(one tick per video frame is typically not sufficient). The clock

frequency is dependent on the format of data carried as payload

and is specified statically in the profile or payload format

specification that defines the format, or MAY be specified

dynamically for payload formats defined through non-RTP means. If

RTP packets are generated periodically, the nominal sampling

instant as determined from the sampling clock is to be used, not a

reading of the system clock. As an example, for fixed-rate audio

the timestamp clock would likely increment by one for each

sampling period. If an audio application reads blocks covering

Schulzrinne, et al. Standards Track [Page 14]

RFC 3550 RTP July 2003

160 sampling periods from the input device, the timestamp would be

increased by 160 for each such block, regardless of whether the

block is transmitted in a packet or dropped as silent.

The initial value of the timestamp SHOULD be random, as for the

sequence number. Several consecutive RTP packets will have equal

timestamps if they are (logically) generated at once, e.g., belong

to the same video frame. Consecutive RTP packets MAY contain

timestamps that are not monotonic if the data is not transmitted

in the order it was sampled, as in the case of MPEG interpolated

video frames. (The sequence numbers of the packets as transmitted

will still be monotonic.)

RTP timestamps from different media streams may advance at

different rates and usually have independent, random offsets.

Therefore, although these timestamps are sufficient to reconstruct

the timing of a single stream, directly comparing RTP timestamps

from different media is not effective for synchronization.

Instead, for each medium the RTP timestamp is related to the

sampling instant by pairing it with a timestamp from a reference

clock (wallclock) that represents the time when the data

corresponding to the RTP timestamp was sampled. The reference

clock is shared by all media to be synchronized. The timestamp

pairs are not transmitted in every data packet, but at a lower

rate in RTCP SR packets as described in Section 6.4.

The sampling instant is chosen as the point of reference for the

RTP timestamp because it is known to the transmitting endpoint and

has a common definition for all media, independent of encoding

delays or other processing. The purpose is to allow synchronized

presentation of all media sampled at the same time.

Applications transmitting stored data rather than data sampled in

real time typically use a virtual presentation timeline derived

from wallclock time to determine when the next frame or other unit

of each medium in the stored data should be presented. In this

case, the RTP timestamp would reflect the presentation time for

each unit. That is, the RTP timestamp for each unit would be

related to the wallclock time at which the unit becomes current on

the virtual presentation timeline. Actual presentation occurs

some time later as determined by the receiver.

An example describing live audio narration of prerecorded video

illustrates the significance of choosing the sampling instant as

the reference point. In this scenario, the video would be

presented locally for the narrator to view and would be

simultaneously transmitted using RTP. The "sampling instant" of a

video frame transmitted in RTP would be established by referencing

Schulzrinne, et al. Standards Track [Page 15]

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its timestamp to the wallclock time when that video frame was

presented to the narrator. The sampling instant for the audio RTP

packets containing the narrator's speech would be established by

referencing the same wallclock time when the audio was sampled.

The audio and video may even be transmitted by different hosts if

the reference clocks on the two hosts are synchronized by some

means such as NTP. A receiver can then synchronize presentation

of the audio and video packets by relating their RTP timestamps

using the timestamp pairs in RTCP SR packets.

SSRC: 32 bits

The SSRC field identifies the synchronization source. This

identifier SHOULD be chosen randomly, with the intent that no two

synchronization sources within the same RTP session will have the

same SSRC identifier. An example algorithm for generating a

random identifier is presented in Appendix A.6. Although the

probability of multiple sources choosing the same identifier is

low, all RTP implementations must be prepared to detect and

resolve collisions. Section 8 describes the probability of

collision along with a mechanism for resolving collisions and

detecting RTP-level forwarding loops based on the uniqueness of

the SSRC identifier. If a source changes its source transport

address, it must also choose a new SSRC identifier to avoid being

interpreted as a looped source (see Section 8.2).

CSRC list: 0 to 15 items, 32 bits each

The CSRC list identifies the contributing sources for the payload

contained in this packet. The number of identifiers is given by

the CC field. If there are more than 15 contributing sources,

only 15 can be identified. CSRC identifiers are inserted by

mixers (see Section 7.1), using the SSRC identifiers of

contributing sources. For example, for audio packets the SSRC

identifiers of all sources that were mixed together to create a

packet are listed, allowing correct talker indication at the

receiver.

5.2 Multiplexing RTP Sessions

For efficient protocol processing, the number of multiplexing points

should be minimized, as described in the integrated layer processing

design principle [10]. In RTP, multiplexing is provided by the

destination transport address (network address and port number) which

is different for each RTP session. For example, in a teleconference

composed of audio and video media encoded separately, each medium

SHOULD be carried in a separate RTP session with its own destination

transport address.

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Separate audio and video streams SHOULD NOT be carried in a single

RTP session and demultiplexed based on the payload type or SSRC

fields. Interleaving packets with different RTP media types but

using the same SSRC would introduce several problems:

1. If, say, two audio streams shared the same RTP session and the

same SSRC value, and one were to change encodings and thus acquire

a different RTP payload type, there would be no general way of

identifying which stream had changed encodings.

2. An SSRC is defined to identify a single timing and sequence number

space. Interleaving multiple payload types would require

different timing spaces if the media clock rates differ and would

require different sequence number spaces to tell which payload

type suffered packet loss.

3. The RTCP sender and receiver reports (see Section 6.4) can only

describe one timing and sequence number space per SSRC and do not

carry a payload type field.

4. An RTP mixer would not be able to combine interleaved streams of

incompatible media into one stream.

5. Carrying multiple media in one RTP session precludes: the use of

different network paths or network resource allocations if

appropriate; reception of a subset of the media if desired, for

example just audio if video would exceed the available bandwidth;

and receiver implementations that use separate processes for the

different media, whereas using separate RTP sessions permits

either single- or multiple-process implementations.

Using a different SSRC for each medium but sending them in the same

RTP session would avoid the first three problems but not the last

two.

On the other hand, multiplexing multiple related sources of the same

medium in one RTP session using different SSRC values is the norm for

multicast sessions. The problems listed above don't apply: an RTP

mixer can combine multiple audio sources, for example, and the same

treatment is applicable for all of them. It may also be appropriate

to multiplex streams of the same medium using different SSRC values

in other scenarios where the last two problems do not apply.

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5.3 Profile-Specific Modifications to the RTP Header

The existing RTP data packet header is believed to be complete for

the set of functions required in common across all the application

classes that RTP might support. However, in keeping with the ALF

design principle, the header MAY be tailored through modifications or

additions defined in a profile specification while still allowing

profile-independent monitoring and recording tools to function.

o The marker bit and payload type field carry profile-specific

information, but they are allocated in the fixed header since many

applications are expected to need them and might otherwise have to

add another 32-bit word just to hold them. The octet containing

these fields MAY be redefined by a profile to suit different

requirements, for example with more or fewer marker bits. If

there are any marker bits, one SHOULD be located in the most

significant bit of the octet since profile-independent monitors

may be able to observe a correlation between packet loss patterns

and the marker bit.

o Additional information that is required for a particular payload

format, such as a video encoding, SHOULD be carried in the payload

section of the packet. This might be in a header that is always

present at the start of the payload section, or might be indicated

by a reserved value in the data pattern.

o If a particular class of applications needs additional

functionality independent of payload format, the profile under

which those applications operate SHOULD define additional fixed

fields to follow immediately after the SSRC field of the existing

fixed header. Those applications will be able to quickly and

directly access the additional fields while profile-independent

monitors or recorders can still process the RTP packets by

interpreting only the first twelve octets.

If it turns out that additional functionality is needed in common

across all profiles, then a new version of RTP should be defined to

make a permanent change to the fixed header.

5.3.1 RTP Header Extension

An extension mechanism is provided to allow individual

implementations to experiment with new payload-format-independent

functions that require additional information to be carried in the

RTP data packet header. This mechanism is designed so that the

header extension may be ignored by other interoperating

implementations that have not been extended.

Schulzrinne, et al. Standards Track [Page 18]

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Note that this header extension is intended only for limited use.

Most potential uses of this mechanism would be better done another

way, using the methods described in the previous section. For

example, a profile-specific extension to the fixed header is less

expensive to process because it is not conditional nor in a variable

location. Additional information required for a particular payload

format SHOULD NOT use this header extension, but SHOULD be carried in

the payload section of the packet.

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| defined by profile | length |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| header extension |

| .... |

If the X bit in the RTP header is one, a variable-length header

extension MUST be appended to the RTP header, following the CSRC list

if present. The header extension contains a 16-bit length field that

counts the number of 32-bit words in the extension, excluding the

four-octet extension header (therefore zero is a valid length). Only

a single extension can be appended to the RTP data header. To allow

multiple interoperating implementations to each experiment

independently with different header extensions, or to allow a

particular implementation to experiment with more than one type of

header extension, the first 16 bits of the header extension are left

open for distinguishing identifiers or parameters. The format of

these 16 bits is to be defined by the profile specification under

which the implementations are operating. This RTP specification does

not define any header extensions itself.

6. RTP Control Protocol -- RTCP

The RTP control protocol (RTCP) is based on the periodic transmission

of control packets to all participants in the session, using the same

distribution mechanism as the data packets. The underlying protocol

MUST provide multiplexing of the data and control packets, for

example using separate port numbers with UDP. RTCP performs four

functions:

1. The primary function is to provide feedback on the quality of the

data distribution. This is an integral part of the RTP's role as

a transport protocol and is related to the flow and congestion

control functions of other transport protocols (see Section 10 on

the requirement for congestion control). The feedback may be

directly useful for control of adaptive encodings [18,19], but

experiments with IP multicasting have shown that it is also

Schulzrinne, et al. Standards Track [Page 19]

RFC 3550 RTP July 2003

critical to get feedback from the receivers to diagnose faults in

the distribution. Sending reception feedback reports to all

participants allows one who is observing problems to evaluate

whether those problems are local or global. With a distribution

mechanism like IP multicast, it is also possible for an entity

such as a network service provider who is not otherwise involved

in the session to receive the feedback information and act as a

third-party monitor to diagnose network problems. This feedback

function is performed by the RTCP sender and receiver reports,

described below in Section 6.4.

2. RTCP carries a persistent transport-level identifier for an RTP

source called the canonical name or CNAME, Section 6.5.1. Since

the SSRC identifier may change if a conflict is discovered or a

program is restarted, receivers require the CNAME to keep track of

each participant. Receivers may also require the CNAME to

associate multiple data streams from a given participant in a set

of related RTP sessions, for example to synchronize audio and

video. Inter-media synchronization also requires the NTP and RTP

timestamps included in RTCP packets by data senders.

3. The first two functions require that all participants send RTCP

packets, therefore the rate must be controlled in order for RTP to

scale up to a large number of participants. By having each

participant send its control packets to all the others, each can

independently observe the number of participants. This number is

used to calculate the rate at which the packets are sent, as

explained in Section 6.2.

4. A fourth, OPTIONAL function is to convey minimal session control

information, for example participant identification to be

displayed in the user interface. This is most likely to be useful

in "loosely controlled" sessions where participants enter and

leave without membership control or parameter negotiation. RTCP

serves as a convenient channel to reach all the participants, but

it is not necessarily expected to support all the control

communication requirements of an application. A higher-level

session control protocol, which is beyond the scope of this

document, may be needed.

Functions 1-3 SHOULD be used in all environments, but particularly in

the IP multicast environment. RTP application designers SHOULD avoid

mechanisms that can only work in unicast mode and will not scale to

larger numbers. Transmission of RTCP MAY be controlled separately

for senders and receivers, as described in Section 6.2, for cases

such as unidirectional links where feedback from receivers is not

possible.

Schulzrinne, et al. Standards Track [Page 20]

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Non-normative note: In the multicast routing approach

called Source-Specific Multicast (SSM), there is only one sender

per "channel" (a source address, group address pair), and

receivers (except for the channel source) cannot use multicast to

communicate directly with other channel members. The

recommendations here accommodate SSM only through Section 6.2's

option of turning off receivers' RTCP entirely. Future work will

specify adaptation of RTCP for SSM so that feedback from receivers

can be maintained.

6.1 RTCP Packet Format

This specification defines several RTCP packet types to carry a

variety of control information:

SR: Sender report, for transmission and reception statistics from

participants that are active senders

RR: Receiver report, for reception statistics from participants

that are not active senders and in combination with SR for

active senders reporting on more than 31 sources

SDES: Source description items, including CNAME

BYE: Indicates end of participation

APP: Application-specific functions

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 MUST end on a 32-bit

boundary. The alignment requirement and a length field in the fixed

part of each packet are included to make RTCP packets "stackable".

Multiple RTCP packets can be concatenated without any intervening

separators to 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.

Each individual RTCP packet in the compound packet may be processed

independently with no requirements upon the order or combination of

packets. However, in order to perform the functions of the protocol,

the following constraints are imposed:

Schulzrinne, et al. Standards Track [Page 21]

RFC 3550 RTP July 2003

o Reception statistics (in SR or RR) should be sent as often as

bandwidth constraints will allow to maximize the resolution of the

statistics, therefore each periodically transmitted compound RTCP

packet MUST include a report packet.

o New receivers need to receive the CNAME for a source as soon as

possible to identify the source and to begin associating media for

purposes such as lip-sync, so each compound RTCP packet MUST also

include the SDES CNAME except when the compound RTCP packet is

split for partial encryption as described in Section 9.1.

o The number of packet types that may appear first in the compound

packet needs to be limited to increase the number of constant bits

in the first word and the probability of successfully validating

RTCP packets against misaddressed RTP data packets or other

unrelated packets.

Thus, all RTCP packets MUST be sent in a compound packet of at least

two individual packets, with the following format:

Encryption prefix: If and only if the compound packet is to be

encrypted according to the method in Section 9.1, it MUST be

prefixed by a random 32-bit quantity redrawn for every compound

packet transmitted. If padding is required for the encryption, it

MUST be added to the last packet of the compound packet.

SR or RR: The first RTCP packet in the compound packet MUST

always be a report packet to facilitate header validation as

described in Appendix A.2. This is true even if no data has been

sent or received, in which case an empty RR MUST be sent, and even

if the only other RTCP packet in the compound packet is a BYE.

Additional RRs: If the number of sources for which reception

statistics are being reported exceeds 31, the number that will fit

into one SR or RR packet, then additional RR packets SHOULD follow

the initial report packet.

SDES: An SDES packet containing a CNAME item MUST be included

in each compound RTCP packet, except as noted in Section 9.1.

Other source description items MAY optionally be included if

required by a particular application, subject to bandwidth

constraints (see Section 6.3.9).

BYE or APP: Other RTCP packet types, including those yet to be

defined, MAY follow in any order, except that BYE SHOULD be the

last packet sent with a given SSRC/CSRC. Packet types MAY appear

more than once.

Schulzrinne, et al. Standards Track [Page 22]

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An individual RTP participant SHOULD send only one compound RTCP

packet per report interval in order for the RTCP bandwidth per

participant to be estimated correctly (see Section 6.2), except when

the compound RTCP packet is split for partial encryption as described

in Section 9.1. If there are too many sources to fit all the

necessary RR packets into one compound RTCP packet without exceeding

the maximum transmission unit (MTU) of the network path, then only

the subset that will fit into one MTU SHOULD be included in each

interval. The subsets SHOULD be selected round-robin across multiple

intervals so that all sources are reported.

It is RECOMMENDED that translators and mixers combine individual RTCP

packets from the multiple sources they are forwarding into one

compound packet whenever feasible in order to amortize the packet

overhead (see Section 7). An example RTCP compound packet as might

be produced by a mixer is shown in Fig. 1. If the overall length of

a compound packet would exceed the MTU of the network path, it SHOULD

be segmented into multiple shorter compound packets to be transmitted

in separate packets of the underlying protocol. This does not impair

the RTCP bandwidth estimation because each compound packet represents

at least one distinct participant. Note that each of the compound

packets MUST begin with an SR or RR packet.

An implementation SHOULD ignore incoming RTCP packets with types

unknown to it. Additional RTCP packet types may be registered with

the Internet Assigned Numbers Authority (IANA) as described in

Section 15.

if encrypted: random 32-bit integer

|

|[--------- packet --------][---------- packet ----------][-packet-]

|

| receiver chunk chunk

V reports item item item item

--------------------------------------------------------------------

R[SR #sendinfo #site1#site2][SDES #CNAME PHONE #CNAME LOC][BYE##why]

--------------------------------------------------------------------

| |

|<----------------------- compound packet ----------------------->|

|<-------------------------- UDP packet ------------------------->|

#: SSRC/CSRC identifier

Figure 1: Example of an RTCP compound packet

Schulzrinne, et al. Standards Track [Page 23]

RFC 3550 RTP July 2003

6.2 RTCP Transmission Interval

RTP is designed to allow an application to scale automatically over

session sizes ranging from a few participants to thousands. For

example, in an audio conference the data traffic is inherently selflimiting

because only one or two people will speak at a time, so with

multicast distribution the data rate on any given link remains

relatively constant independent of the number of participants.

However, the control traffic is not self-limiting. If the reception

reports from each participant were sent at a constant rate, the

control traffic would grow linearly with the number of participants.

Therefore, the rate must be scaled down by dynamically calculating

the interval between RTCP packet transmissions.

For each session, it is assumed that the data traffic is subject to

an aggregate limit called the "session bandwidth" to be divided among

the participants. This bandwidth might be reserved and the limit

enforced by the network. If there is no reservation, there may be

other constraints, depending on the environment, that establish the

"reasonable" maximum for the session to use, and that would be the

session bandwidth. The session bandwidth may be chosen based on some

cost or a priori knowledge of the available network bandwidth for the

session. It is somewhat independent of the media encoding, but the

encoding choice may be limited by the session bandwidth. Often, the

session bandwidth is the sum of the nominal bandwidths of the senders

expected to be concurrently active. For teleconference audio, this

number would typically be one sender's bandwidth. For layered

encodings, each layer is a separate RTP session with its own session

bandwidth parameter.

The session bandwidth parameter is expected to be supplied by a

session management application when it invokes a media application,

but media applications MAY set a default based on the single-sender

data bandwidth for the encoding selected for the session. The

application MAY also enforce bandwidth limits based on multicast

scope rules or other criteria. All participants MUST use the same

value for the session bandwidth so that the same RTCP interval will

be calculated.

Bandwidth calculations for control and data traffic include lowerlayer

transport and network protocols (e.g., UDP and IP) since that

is what the resource reservation system would need to know. The

application can also be expected to know which of these protocols are

in use. Link level headers are not included in the calculation since

the packet will be encapsulated with different link level headers as

it travels.

Schulzrinne, et al. Standards Track [Page 24]

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The control traffic should be limited to a small and known fraction

of the session bandwidth: small so that the primary function of the

transport protocol to carry data is not impaired; known so that the

control traffic can be included in the bandwidth specification given

to a resource reservation protocol, and so that each participant can

independently calculate its share. The control traffic bandwidth is

in addition to the session bandwidth for the data traffic. It is

RECOMMENDED that the fraction of the session bandwidth added for RTCP

be fixed at 5%. It is also RECOMMENDED that 1/4 of the RTCP

bandwidth be dedicated to participants that are sending data so that

in sessions with a large number of receivers but a small number of

senders, newly joining participants will more quickly receive the

CNAME for the sending sites. When the proportion of senders is

greater than 1/4 of the participants, the senders get their

proportion of the full RTCP bandwidth. While the values of these and

other constants in the interval calculation are not critical, all

participants in the session MUST use the same values so the same

interval will be calculated. Therefore, these constants SHOULD be

fixed for a particular profile.

A profile MAY specify that the control traffic bandwidth may be a

separate parameter of the session rather than a strict percentage of

the session bandwidth. Using a separate parameter allows rateadaptive

applications to set an RTCP bandwidth consistent with a

"typical" data bandwidth that is lower than the maximum bandwidth

specified by the session bandwidth parameter.

The profile MAY further specify that the control traffic bandwidth

may be divided into two separate session parameters for those

participants which are active data senders and those which are not;

let us call the parameters S and R. Following the recommendation

that 1/4 of the RTCP bandwidth be dedicated to data senders, the

RECOMMENDED default values for these two parameters would be 1.25%

and 3.75%, respectively. When the proportion of senders is greater

than S/(S+R) of the participants, the senders get their proportion of

the sum of these parameters. Using two parameters allows RTCP

reception reports to be turned off entirely for a particular session

by setting the RTCP bandwidth for non-data-senders to zero while

keeping the RTCP bandwidth for data senders non-zero so that sender

reports can still be sent for inter-media synchronization. Turning

off RTCP reception reports is NOT RECOMMENDED because they are needed

for the functions listed at the beginning of Section 6, particularly

reception quality feedback and congestion control. However, doing so

may be appropriate for systems operating on unidirectional links or

for sessions that don't require feedback on the quality of reception

or liveness of receivers and that have other means to avoid

congestion.

Schulzrinne, et al. Standards Track [Page 25]

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The calculated interval between transmissions of compound RTCP

packets SHOULD also have a lower bound to avoid having bursts of

packets exceed the allowed bandwidth when the number of participants

is small and the traffic isn't smoothed according to the law of large

numbers. It also keeps the report interval from becoming too small

during transient outages like a network partition such that

adaptation is delayed when the partition heals. At application

startup, a delay SHOULD be imposed before the first compound RTCP

packet is sent to allow time for RTCP packets to be received from

other participants so the report interval will converge to the

correct value more quickly. This delay MAY be set to half the

minimum interval to allow quicker notification that the new

participant is present. The RECOMMENDED value for a fixed minimum

interval is 5 seconds.

An implementation MAY scale the minimum RTCP interval to a smaller

value inversely proportional to the session bandwidth parameter with

the following limitations:

o For multicast sessions, only active data senders MAY use the

reduced minimum value to calculate the interval for transmission

of compound RTCP packets.

o For unicast sessions, the reduced value MAY be used by

participants that are not active data senders as well, and the

delay before sending the initial compound RTCP packet MAY be zero.

o For all sessions, the fixed minimum SHOULD be used when

calculating the participant timeout interval (see Section 6.3.5)

so that implementations which do not use the reduced value for

transmitting RTCP packets are not timed out by other participants

prematurely.

o The RECOMMENDED value for the reduced minimum in seconds is 360

divided by the session bandwidth in kilobits/second. This minimum

is smaller than 5 seconds for bandwidths greater than 72 kb/s.

The algorithm described in Section 6.3 and Appendix A.7 was designed

to meet the goals outlined in this section. It calculates the

interval between sending compound RTCP packets to divide the allowed

control traffic bandwidth among the participants. This allows an

application to provide fast response for small sessions where, for

example, identification of all participants is important, yet

automatically adapt to large sessions. The algorithm incorporates

the following characteristics:

Schulzrinne, et al. Standards Track [Page 26]

RFC 3550 RTP July 2003

o The calculated interval between RTCP packets scales linearly with

the number of members in the group. It is this linear factor

which allows for a constant amount of control traffic when summed

across all members.

o The interval between RTCP packets is varied randomly over the

range [0.5,1.5] times the calculated interval to avoid unintended

synchronization of all participants [20]. The first RTCP packet

sent after joining a session is also delayed by a random variation

of half the minimum RTCP interval.

o A dynamic estimate of the average compound RTCP packet size is

calculated, including all those packets received and sent, to

automatically adapt to changes in the amount of control

information carried.

o Since the calculated interval is dependent on the number of

observed group members, there may be undesirable startup effects

when a new user joins an existing session, or many users

simultaneously join a new session. These new users will initially

have incorrect estimates of the group membership, and thus their

RTCP transmission interval will be too short. This problem can be

significant if many users join the session simultaneously. To

deal with this, an algorithm called "timer reconsideration" is

employed. This algorithm implements a simple back-off mechanism

which causes users to hold back RTCP packet transmission if the

group sizes are increasing.

o When users leave a session, either with a BYE or by timeout, the

group membership decreases, and thus the calculated interval

should decrease. A "reverse reconsideration" algorithm is used to

allow members to more quickly reduce their intervals in response

to group membership decreases.

o BYE packets are given different treatment than other RTCP packets.

When a user leaves a group, and wishes to send a BYE packet, it

may do so before its next scheduled RTCP packet. However,

transmission of BYEs follows a back-off algorithm which avoids

floods of BYE packets should a large number of members

simultaneously leave the session.

This algorithm may be used for sessions in which all participants are

allowed to send. In that case, the session bandwidth parameter is

the product of the individual sender's bandwidth times the number of

participants, and the RTCP bandwidth is 5% of that.

Details of the algorithm's operation are given in the sections that

follow. Appendix A.7 gives an example implementation.

Schulzrinne, et al. Standards Track [Page 27]

RFC 3550 RTP July 2003

6.2.1 Maintaining the Number of Session Members

Calculation of the RTCP packet interval depends upon an estimate of

the number of sites participating in the session. New sites are

added to the count when they are heard, and an entry for each SHOULD

be created in a table indexed by the SSRC or CSRC identifier (see

Section 8.2) to keep track of them. New entries MAY be considered

not valid until multiple packets carrying the new SSRC have been

received (see Appendix A.1), or until an SDES RTCP packet containing

a CNAME for that SSRC has been received. Entries MAY be deleted from

the table when an RTCP BYE packet with the corresponding SSRC

identifier is received, except that some straggler data packets might

arrive after the BYE and cause the entry to be recreated. Instead,

the entry SHOULD be marked as having received a BYE and then deleted

after an appropriate delay.

A participant MAY mark another site inactive, or delete it if not yet

valid, if no RTP or RTCP packet has been received for a small number

of RTCP report intervals (5 is RECOMMENDED). This provides some

robustness against packet loss. All sites must have the same value

for this multiplier and must calculate roughly the same value for the

RTCP report interval in order for this timeout to work properly.

Therefore, this multiplier SHOULD be fixed for a particular profile.

For sessions with a very large number of participants, it may be

impractical to maintain a table to store the SSRC identifier and

state information for all of them. An implementation MAY use SSRC

sampling, as described in [21], to reduce the storage requirements.

An implementation MAY use any other algorithm with similar

performance. A key requirement is that any algorithm considered

SHOULD NOT substantially underestimate the group size, although it

MAY overestimate.

6.3 RTCP Packet Send and Receive Rules

The rules for how to send, and what to do when receiving an RTCP

packet are outlined here. An implementation that allows operation in

a multicast environment or a multipoint unicast environment MUST meet

the requirements in Section 6.2. Such an implementation MAY use the

algorithm defined in this section to meet those requirements, or MAY

use some other algorithm so long as it provides equivalent or better

performance. An implementation which is constrained to two-party

unicast operation SHOULD still use randomization of the RTCP

transmission interval to avoid unintended synchronization of multiple

instances operating in the same environment, but MAY omit the "timer

reconsideration" and "reverse reconsideration" algorithms in Sections

6.3.3, 6.3.6 and 6.3.7.

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To execute these rules, a session participant must maintain several

pieces of state:

tp: the last time an RTCP packet was transmitted;

tc: the current time;

tn: the next scheduled transmission time of an RTCP packet;

pmembers: the estimated number of session members at the time tn

was last recomputed;

members: the most current estimate for the number of session

members;

senders: the most current estimate for the number of senders in

the session;

rtcp_bw: The target RTCP bandwidth, i.e., the total bandwidth

that will be used for RTCP packets by all members of this session,

in octets per second. This will be a specified fraction of the

"session bandwidth" parameter supplied to the application at

startup.

we_sent: Flag that is true if the application has sent data

since the 2nd previous RTCP report was transmitted.

avg_rtcp_size: The average compound RTCP packet size, in octets,

over all RTCP packets sent and received by this participant. The

size includes lower-layer transport and network protocol headers

(e.g., UDP and IP) as explained in Section 6.2.

initial: Flag that is true if the application has not yet sent

an RTCP packet.

Many of these rules make use of the "calculated interval" between

packet transmissions. This interval is described in the following

section.

6.3.1 Computing the RTCP Transmission Interval

To maintain scalability, the average interval between packets from a

session participant should scale with the group size. This interval

is called the calculated interval. It is obtained by combining a

number of the pieces of state described above. The calculated

interval T is then determined as follows:

Schulzrinne, et al. Standards Track [Page 29]

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1. If the number of senders is less than or equal to 25% of the

membership (members), the interval depends on whether the

participant is a sender or not (based on the value of we_sent).

If the participant is a sender (we_sent true), the constant C is

set to the average RTCP packet size (avg_rtcp_size) divided by 25%

of the RTCP bandwidth (rtcp_bw), and the constant n is set to the

number of senders. If we_sent is not true, the constant C is set

to the average RTCP packet size divided by 75% of the RTCP

bandwidth. The constant n is set to the number of receivers

(members - senders). If the number of senders is greater than

25%, senders and receivers are treated together. The constant C

is set to the average RTCP packet size divided by the total RTCP

bandwidth and n is set to the total number of members. As stated

in Section 6.2, an RTP profile MAY specify that the RTCP bandwidth

may be explicitly defined by two separate parameters (call them S

and R) for those participants which are senders and those which

are not. In that case, the 25% fraction becomes S/(S+R) and the

75% fraction becomes R/(S+R). Note that if R is zero, the

percentage of senders is never greater than S/(S+R), and the

implementation must avoid division by zero.

2. If the participant has not yet sent an RTCP packet (the variable

initial is true), the constant Tmin is set to 2.5 seconds, else it

is set to 5 seconds.

3. The deterministic calculated interval Td is set to max(Tmin, n*C).

4. The calculated interval T is set to a number uniformly distributed

between 0.5 and 1.5 times the deterministic calculated interval.

5. The resulting value of T is divided by e-3/2=1.21828 to compensate

for the fact that the timer reconsideration algorithm converges to

a value of the RTCP bandwidth below the intended average.

This procedure results in an interval which is random, but which, on

average, gives at least 25% of the RTCP bandwidth to senders and the

rest to receivers. If the senders constitute more than one quarter

of the membership, this procedure splits the bandwidth equally among

all participants, on average.

6.3.2 Initialization

Upon joining the session, the participant initializes tp to 0, tc to

0, senders to 0, pmembers to 1, members to 1, we_sent to false,

rtcp_bw to the specified fraction of the session bandwidth, initial

to true, and avg_rtcp_size to the probable size of the first RTCP

packet that the application will later construct. The calculated

interval T is then computed, and the first packet is scheduled for

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time tn = T. This means that a transmission timer is set which

expires at time T. Note that an application MAY use any desired

approach for implementing this timer.

The participant adds its own SSRC to the member table.

6.3.3 Receiving an RTP or Non-BYE RTCP Packet

When an RTP or RTCP packet is received from a participant whose SSRC

is not in the member table, the SSRC is added to the table, and the

value for members is updated once the participant has been validated

as described in Section 6.2.1. The same processing occurs for each

CSRC in a validated RTP packet.

When an RTP packet is received from a participant whose SSRC is not

in the sender table, the SSRC is added to the table, and the value

for senders is updated.

For each compound RTCP packet received, the value of avg_rtcp_size is

updated:

avg_rtcp_size = (1/16) * packet_size + (15/16) * avg_rtcp_size

where packet_size is the size of the RTCP packet just received.

6.3.4 Receiving an RTCP BYE Packet

Except as described in Section 6.3.7 for the case when an RTCP BYE is

to be transmitted, if the received packet is an RTCP BYE packet, the

SSRC is checked against the member table. If present, the entry is

removed from the table, and the value for members is updated. The

SSRC is then checked against the sender table. If present, the entry

is removed from the table, and the value for senders is updated.

Furthermore, to make the transmission rate of RTCP packets more

adaptive to changes in group membership, the following "reverse

reconsideration" algorithm SHOULD be executed when a BYE packet is

received that reduces members to a value less than pmembers:

o The value for tn is updated according to the following formula:

tn = tc + (members/pmembers) * (tn - tc)

o The value for tp is updated according the following formula:

tp = tc - (members/pmembers) * (tc - tp).

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o The next RTCP packet is rescheduled for transmission at time tn,

which is now earlier.

o The value of pmembers is set equal to members.

This algorithm does not prevent the group size estimate from

incorrectly dropping to zero for a short time due to premature

timeouts when most participants of a large session leave at once but

some remain. The algorithm does make the estimate return to the

correct value more rapidly. This situation is unusual enough and the

consequences are sufficiently harmless that this problem is deemed

only a secondary concern.

6.3.5 Timing Out an SSRC

At occasional intervals, the participant MUST check to see if any of

the other participants time out. To do this, the participant

computes the deterministic (without the randomization factor)

calculated interval Td for a receiver, that is, with we_sent false.

Any other session member who has not sent an RTP or RTCP packet since

time tc - MTd (M is the timeout multiplier, and defaults to 5) is

timed out. This means that its SSRC is removed from the member list,

and members is updated. A similar check is performed on the sender

list. Any member on the sender list who has not sent an RTP packet

since time tc - 2T (within the last two RTCP report intervals) is

removed from the sender list, and senders is updated.

If any members time out, the reverse reconsideration algorithm

described in Section 6.3.4 SHOULD be performed.

The participant MUST perform this check at least once per RTCP

transmission interval.

6.3.6 Expiration of Transmission Timer

When the packet transmission timer expires, the participant performs

the following operations:

o The transmission interval T is computed as described in Section

6.3.1, including the randomization factor.

o If tp + T is less than or equal to tc, an RTCP packet is

transmitted. tp is set to tc, then another value for T is

calculated as in the previous step and tn is set to tc + T. The

transmission timer is set to expire again at time tn. If tp + T

is greater than tc, tn is set to tp + T. No RTCP packet is

transmitted. The transmission timer is set to expire at time tn.

Schulzrinne, et al. Standards Track [Page 32]

RFC 3550 RTP July 2003

o pmembers is set to members.

If an RTCP packet is transmitted, the value of initial is set to

FALSE. Furthermore, the value of avg_rtcp_size is updated:

avg_rtcp_size = (1/16) * packet_size + (15/16) * avg_rtcp_size

where packet_size is the size of the RTCP packet just transmitted.

6.3.7 Transmitting a BYE Packet

When a participant wishes to leave a session, a BYE packet is

transmitted to inform the other participants of the event. In order

to avoid a flood of BYE packets when many participants leave the

system, a participant MUST execute the following algorithm if the

number of members is more than 50 when the participant chooses to

leave. This algorithm usurps the normal role of the members variable

to count BYE packets instead:

o When the participant decides to leave the system, tp is reset to

tc, the current time, members and pmembers are initialized to 1,

initial is set to 1, we_sent is set to false, senders is set to 0,

and avg_rtcp_size is set to the size of the compound BYE packet.

The calculated interval T is computed. The BYE packet is then

scheduled for time tn = tc + T.

o Every time a BYE packet from another participant is received,

members is incremented by 1 regardless of whether that participant

exists in the member table or not, and when SSRC sampling is in

use, regardless of whether or not the BYE SSRC would be included

in the sample. members is NOT incremented when other RTCP packets

or RTP packets are received, but only for BYE packets. Similarly,

avg_rtcp_size is updated only for received BYE packets. senders

is NOT updated when RTP packets arrive; it remains 0.

o Transmission of the BYE packet then follows the rules for

transmitting a regular RTCP packet, as above.

This allows BYE packets to be sent right away, yet controls their

total bandwidth usage. In the worst case, this could cause RTCP

control packets to use twice the bandwidth as normal (10%) -- 5% for

non-BYE RTCP packets and 5% for BYE.

A participant that does not want to wait for the above mechanism to

allow transmission of a BYE packet MAY leave the group without

sending a BYE at all. That participant will eventually be timed out

by the other group members.

Schulzrinne, et al. Standards Track [Page 33]

RFC 3550 RTP July 2003

If the group size estimate members is less than 50 when the

participant decides to leave, the participant MAY send a BYE packet

immediately. Alternatively, the participant MAY choose to execute

the above BYE backoff algorithm.

In either case, a participant which never sent an RTP or RTCP packet

MUST NOT send a BYE packet when they leave the group.

6.3.8 Updating we_sent

The variable we_sent contains true if the participant has sent an RTP

packet recently, false otherwise. This determination is made by

using the same mechanisms as for managing the set of other

participants listed in the senders table. If the participant sends

an RTP packet when we_sent is false, it adds itself to the sender

table and sets we_sent to true. The reverse reconsideration

algorithm described in Section 6.3.4 SHOULD be performed to possibly

reduce the delay before sending an SR packet. Every time another RTP

packet is sent, the time of transmission of that packet is maintained

in the table. The normal sender timeout algorithm is then applied to

the participant -- if an RTP packet has not been transmitted since

time tc - 2T, the participant removes itself from the sender table,

decrements the sender count, and sets we_sent to false.

6.3.9 Allocation of Source Description Bandwidth

This specification defines several source description (SDES) items in

addition to the mandatory CNAME item, such as NAME (personal name)

and EMAIL (email address). It also provides a means to define new

application-specific RTCP packet types. Applications should exercise

caution in allocating control bandwidth to this additional

information because it will slow down the rate at which reception

reports and CNAME are sent, thus impairing the performance of the

protocol. It is RECOMMENDED that no more than 20% of the RTCP

bandwidth allocated to a single participant be used to carry the

additional information. Furthermore, it is not intended that all

SDES items will be included in every application. Those that are

included SHOULD be assigned a fraction of the bandwidth according to

their utility. Rather than estimate these fractions dynamically, it

is recommended that the percentages be translated statically into

report interval counts based on the typical length of an item.

For example, an application may be designed to send only CNAME, NAME

and EMAIL and not any others. NAME might be given much higher

priority than EMAIL because the NAME would be displayed continuously

in the application's user interface, whereas EMAIL would be displayed

only when requested. At every RTCP interval, an RR packet and an

SDES packet with the CNAME item would be sent. For a small session

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RFC 3550 RTP July 2003

operating at the minimum interval, that would be every 5 seconds on

the average. Every third interval (15 seconds), one extra item would

be included in the SDES packet. Seven out of eight times this would

be the NAME item, and every eighth time (2 minutes) it would be the

EMAIL item.

When multiple applications operate in concert using cross-application

binding through a common CNAME for each participant, for example in a

multimedia conference composed of an RTP session for each medium, the

additional SDES information MAY be sent in only one RTP session. The

other sessions would carry only the CNAME item. In particular, this

approach should be applied to the multiple sessions of a layered

encoding scheme (see Section 2.4).

6.4 Sender and Receiver Reports

RTP receivers provide reception quality feedback using RTCP report

packets which may take one of two forms depending upon whether or not

the receiver is also a sender. The only difference between the

sender report (SR) and receiver report (RR) forms, besides the packet

type code, is that the sender report includes a 20-byte sender

information section for use by active senders. The SR is issued if a

site has sent any data packets during the interval since issuing the

last report or the previous one, otherwise the RR is issued.

Both the SR and RR forms include zero or more reception report

blocks, one for each of the synchronization sources from which this

receiver has received RTP data packets since the last report.

Reports are not issued for contributing sources listed in the CSRC

list. Each reception report block provides statistics about the data

received from the particular source indicated in that block. Since a

maximum of 31 reception report blocks will fit in an SR or RR packet,

additional RR packets SHOULD be stacked after the initial SR or RR

packet as needed to contain the reception reports for all sources

heard during the interval since the last report. If there are too

many sources to fit all the necessary RR packets into one compound

RTCP packet without exceeding the MTU of the network path, then only

the subset that will fit into one MTU SHOULD be included in each

interval. The subsets SHOULD be selected round-robin across multiple

intervals so that all sources are reported.

The next sections define the formats of the two reports, how they may

be extended in a profile-specific manner if an application requires

additional feedback information, and how the reports may be used.

Details of reception reporting by translators and mixers is given in

Section 7.

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RFC 3550 RTP July 2003

6.4.1 SR: Sender Report RTCP Packet

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

header |V=2|P| RC | PT=SR=200 | length |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| SSRC of sender |

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

sender | NTP timestamp, most significant word |

info +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| NTP timestamp, least significant word |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| RTP timestamp |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| sender's packet count |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| sender's octet count |

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

report | SSRC_1 (SSRC of first source) |

block +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

1 | fraction lost | cumulative number of packets lost |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| extended highest sequence number received |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| interarrival jitter |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| last SR (LSR) |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| delay since last SR (DLSR) |

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

report | SSRC_2 (SSRC of second source) |

block +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

2 : ... :

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

| profile-specific extensions |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The sender report packet consists of three sections, possibly

followed by a fourth profile-specific extension section if defined.

The first section, the header, is 8 octets long. The fields have the

following meaning:

version (V): 2 bits

Identifies the version of RTP, which is the same in RTCP packets

as in RTP data packets. The version defined by this specification

is two (2).

Schulzrinne, et al. Standards Track [Page 36]

RFC 3550 RTP July 2003

padding (P): 1 bit

If the padding bit is set, this individual RTCP packet contains

some additional padding octets at the end which are not part of

the control information but are included in the length field. The

last octet of the padding is a count of how many padding octets

should be ignored, including itself (it will be a multiple of

four). Padding may be needed by some encryption algorithms with

fixed block sizes. In a compound RTCP packet, padding is only

required on one individual packet because the compound packet is

encrypted as a whole for the method in Section 9.1. Thus, padding

MUST only be added to the last individual packet, and if padding

is added to that packet, the padding bit MUST be set only on that

packet. This convention aids the header validity checks described

in Appendix A.2 and allows detection of packets from some early

implementations that incorrectly set the padding bit on the first

individual packet and add padding to the last individual packet.

reception report count (RC): 5 bits

The number of reception report blocks contained in this packet. A

value of zero is valid.

packet type (PT): 8 bits

Contains the constant 200 to identify this as an RTCP SR packet.

length: 16 bits

The length of this RTCP packet in 32-bit words minus one,

including the header and any padding. (The offset of one makes

zero a valid length and avoids a possible infinite loop in

scanning a compound RTCP packet, while counting 32-bit words

avoids a validity check for a multiple of 4.)

SSRC: 32 bits

The synchronization source identifier for the originator of this

SR packet.

The second section, the sender information, is 20 octets long and is

present in every sender report packet. It summarizes the data

transmissions from this sender. The fields have the following

meaning:

NTP timestamp: 64 bits

Indicates the wallclock time (see Section 4) when this report was

sent so that it may be used in combination with timestamps

returned in reception reports from other receivers to measure

round-trip propagation to those receivers. Receivers should

expect that the measurement accuracy of the timestamp may be

limited to far less than the resolution of the NTP timestamp. The

measurement uncertainty of the timestamp is not indicated as it

Schulzrinne, et al. Standards Track [Page 37]

RFC 3550 RTP July 2003

may not be known. On a system that has no notion of wallclock

time but does have some system-specific clock such as "system

uptime", a sender MAY use that clock as a reference to calculate

relative NTP timestamps. It is important to choose a commonly

used clock so that if separate implementations are used to produce

the individual streams of a multimedia session, all

implementations will use the same clock. Until the year 2036,

relative and absolute timestamps will differ in the high bit so

(invalid) comparisons will show a large difference; by then one

hopes relative timestamps will no longer be needed. A sender that

has no notion of wallclock or elapsed time MAY set the NTP

timestamp to zero.

RTP timestamp: 32 bits

Corresponds to the same time as the NTP timestamp (above), but in

the same units and with the same random offset as the RTP

timestamps in data packets. This correspondence may be used for

intra- and inter-media synchronization for sources whose NTP

timestamps are synchronized, and may be used by media-independent

receivers to estimate the nominal RTP clock frequency. Note that

in most cases this timestamp will not be equal to the RTP

timestamp in any adjacent data packet. Rather, it MUST be

calculated from the corresponding NTP timestamp using the

relationship between the RTP timestamp counter and real time as

maintained by periodically checking the wallclock time at a

sampling instant.

sender's packet count: 32 bits

The total number of RTP data packets transmitted by the sender

since starting transmission up until the time this SR packet was

generated. The count SHOULD be reset if the sender changes its

SSRC identifier.

sender's octet count: 32 bits

The total number of payload octets (i.e., not including header or

padding) transmitted in RTP data packets by the sender since

starting transmission up until the time this SR packet was

generated. The count SHOULD be reset if the sender changes its

SSRC identifier. This field can be used to estimate the average

payload data rate.

The third section contains zero or more reception report blocks

depending on the number of other sources heard by this sender since

the last report. Each reception report block conveys statistics on

the reception of RTP packets from a single synchronization source.

Receivers SHOULD NOT carry over statistics when a source changes its

SSRC identifier due to a collision. These statistics are:

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RFC 3550 RTP July 2003

SSRC_n (source identifier): 32 bits

The SSRC identifier of the source to which the information in this

reception report block pertains.

fraction lost: 8 bits

The fraction of RTP data packets from source SSRC_n lost since the

previous SR or RR packet was sent, expressed as a fixed point

number with the binary point at the left edge of the field. (That

is equivalent to taking the integer part after multiplying the

loss fraction by 256.) This fraction is defined to be the number

of packets lost divided by the number of packets expected, as

defined in the next paragraph. An implementation is shown in

Appendix A.3. If the loss is negative due to duplicates, the

fraction lost is set to zero. Note that a receiver cannot tell

whether any packets were lost after the last one received, and

that there will be no reception report block issued for a source

if all packets from that source sent during the last reporting

interval have been lost.

cumulative number of packets lost: 24 bits

The total number of RTP data packets from source SSRC_n that have

been lost since the beginning of reception. This number is

defined to be the number of packets expected less the number of

packets actually received, where the number of packets received

includes any which are late or duplicates. Thus, packets that

arrive late are not counted as lost, and the loss may be negative

if there are duplicates. The number of packets expected is

defined to be the extended last sequence number received, as

defined next, less the initial sequence number received. This may

be calculated as shown in Appendix A.3.

extended highest sequence number received: 32 bits

The low 16 bits contain the highest sequence number received in an

RTP data packet from source SSRC_n, and the most significant 16

bits extend that sequence number with the corresponding count of

sequence number cycles, which may be maintained according to the

algorithm in Appendix A.1. Note that different receivers within

the same session will generate different extensions to the

sequence number if their start times differ significantly.

interarrival jitter: 32 bits

An estimate of the statistical variance of the RTP data packet

interarrival time, measured in timestamp units and expressed as an

unsigned integer. The interarrival jitter J is defined to be the

mean deviation (smoothed absolute value) of the difference D in

packet spacing at the receiver compared to the sender for a pair

of packets. As shown in the equation below, this is equivalent to

the difference in the "relative transit time" for the two packets;

Schulzrinne, et al. Standards Track [Page 39]

RFC 3550 RTP July 2003

the relative transit time is the difference between a packet's RTP

timestamp and the receiver's clock at the time of arrival,

measured in the same units.

If Si is the RTP timestamp from packet i, and Ri is the time of

arrival in RTP timestamp units for packet i, then for two packets

i and j, D may be expressed as

D(i,j) = (Rj - Ri) - (Sj - Si) = (Rj - Sj) - (Ri - Si)

The interarrival jitter SHOULD be calculated continuously as each

data packet i is received from source SSRC_n, using this

difference D for that packet and the previous packet i-1 in order

of arrival (not necessarily in sequence), according to the formula

J(i) = J(i-1) + (|D(i-1,i)| - J(i-1))/16

Whenever a reception report is issued, the current value of J is

sampled.

The jitter calculation MUST conform to the formula specified here

in order to allow profile-independent monitors to make valid

interpretations of reports coming from different implementations.

This algorithm is the optimal first-order estimator and the gain

parameter 1/16 gives a good noise reduction ratio while

maintaining a reasonable rate of convergence [22]. A sample

implementation is shown in Appendix A.8. See Section 6.4.4 for a

discussion of the effects of varying packet duration and delay

before transmission.

last SR timestamp (LSR): 32 bits

The middle 32 bits out of 64 in the NTP timestamp (as explained in

Section 4) received as part of the most recent RTCP sender report

(SR) packet from source SSRC_n. If no SR has been received yet,

the field is set to zero.

delay since last SR (DLSR): 32 bits

The delay, expressed in units of 1/65536 seconds, between

receiving the last SR packet from source SSRC_n and sending this

reception report block. If no SR packet has been received yet

from SSRC_n, the DLSR field is set to zero.

Let SSRC_r denote the receiver issuing this receiver report.

Source SSRC_n can compute the round-trip propagation delay to

SSRC_r by recording the time A when this reception report block is

received. It calculates the total round-trip time A-LSR using the

last SR timestamp (LSR) field, and then subtracting this field to

leave the round-trip propagation delay as (A - LSR - DLSR). This

Schulzrinne, et al. Standards Track [Page 40]

RFC 3550 RTP July 2003

is illustrated in Fig. 2. Times are shown in both a hexadecimal

representation of the 32-bit fields and the equivalent floatingpoint

decimal representation. Colons indicate a 32-bit field

divided into a 16-bit integer part and 16-bit fraction part.

This may be used as an approximate measure of distance to cluster

receivers, although some links have very asymmetric delays.

[10 Nov 1995 11:33:25.125 UTC] [10 Nov 1995 11:33:36.5 UTC]

n SR(n) A=b710:8000 (46864.500 s)

---------------------------------------------------------------->

v ^

ntp_sec =0xb44db705 v ^ dlsr=0x0005:4000 ( 5.250s)

ntp_frac=0x20000000 v ^ lsr =0xb705:2000 (46853.125s)

(3024992005.125 s) v ^

r v ^ RR(n)

---------------------------------------------------------------->

|<-DLSR->|

(5.250 s)

A 0xb710:8000 (46864.500 s)

DLSR -0x0005:4000 ( 5.250 s)

LSR -0xb705:2000 (46853.125 s)

-------------------------------

delay 0x0006:2000 ( 6.125 s)

Figure 2: Example for round-trip time computation

Schulzrinne, et al. Standards Track [Page 41]

RFC 3550 RTP July 2003

6.4.2 RR: Receiver Report RTCP Packet

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

header |V=2|P| RC | PT=RR=201 | length |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| SSRC of packet sender |

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

report | SSRC_1 (SSRC of first source) |

block +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

1 | fraction lost | cumulative number of packets lost |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| extended highest sequence number received |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| interarrival jitter |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| last SR (LSR) |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| delay since last SR (DLSR) |

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

report | SSRC_2 (SSRC of second source) |

block +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

2 : ... :

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

| profile-specific extensions |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The format of the receiver report (RR) packet is the same as that of

the SR packet except that the packet type field contains the constant

201 and the five words of sender information are omitted (these are

the NTP and RTP timestamps and sender's packet and octet counts).

The remaining fields have the same meaning as for the SR packet.

An empty RR packet (RC = 0) MUST be put at the head of a compound

RTCP packet when there is no data transmission or reception to

report.

6.4.3 Extending the Sender and Receiver Reports

A profile SHOULD define profile-specific extensions to the sender

report and receiver report if there is additional information that

needs to be reported regularly about the sender or receivers. This

method SHOULD be used in preference to defining another RTCP packet

type because it requires less overhead:

o fewer octets in the packet (no RTCP header or SSRC field);

Schulzrinne, et al. Standards Track [Page 42]

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o simpler and faster parsing because applications running under that

profile would be programmed to always expect the extension fields

in the directly accessible location after the reception reports.

The extension is a fourth section in the sender- or receiver-report

packet which comes at the end after the reception report blocks, if

any. If additional sender information is required, then for sender

reports it would be included first in the extension section, but for

receiver reports it would not be present. If information about

receivers is to be included, that data SHOULD be structured as an

array of blocks parallel to the existing array of reception report

blocks; that is, the number of blocks would be indicated by the RC

field.

6.4.4 Analyzing Sender and Receiver Reports

It is expected that reception quality feedback will be useful not

only for the sender but also for other receivers and third-party

monitors. The sender may modify its transmissions based on the

feedback; receivers can determine whether problems are local,

regional or global; network managers may use profile-independent

monitors that receive only the RTCP packets and not the corresponding

RTP data packets to evaluate the performance of their networks for

multicast distribution.

Cumulative counts are used in both the sender information and

receiver report blocks so that differences may be calculated between

any two reports to make measurements over both short and long time

periods, and to provide resilience against the loss of a report. The

difference between the last two reports received can be used to

estimate the recent quality of the distribution. The NTP timestamp

is included so that rates may be calculated from these differences

over the interval between two reports. Since that timestamp is

independent of the clock rate for the data encoding, it is possible

to implement encoding- and profile-independent quality monitors.

An example calculation is the packet loss rate over the interval

between two reception reports. The difference in the cumulative

number of packets lost gives the number lost during that interval.

The difference in the extended last sequence numbers received gives

the number of packets expected during the interval. The ratio of

these two is the packet loss fraction over the interval. This ratio

should equal the fraction lost field if the two reports are

consecutive, but otherwise it may not. The loss rate per second can

be obtained by dividing the loss fraction by the difference in NTP

timestamps, expressed in seconds. The number of packets received is

the number of packets expected minus the number lost. The number of

Schulzrinne, et al. Standards Track [Page 43]

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packets expected may also be used to judge the statistical validity

of any loss estimates. For example, 1 out of 5 packets lost has a

lower significance than 200 out of 1000.

From the sender information, a third-party monitor can calculate the

average payload data rate and the average packet rate over an

interval without receiving the data. Taking the ratio of the two

gives the average payload size. If it can be assumed that packet

loss is independent of packet size, then the number of packets

received by a particular receiver times the average payload size (or

the corresponding packet size) gives the apparent throughput

available to that receiver.

In addition to the cumulative counts which allow long-term packet

loss measurements using differences between reports, the fraction

lost field provides a short-term measurement from a single report.

This becomes more important as the size of a session scales up enough

that reception state information might not be kept for all receivers

or the interval between reports becomes long enough that only one

report might have been received from a particular receiver.

The interarrival jitter field provides a second short-term measure of

network congestion. Packet loss tracks persistent congestion while

the jitter measure tracks transient congestion. The jitter measure

may indicate congestion before it leads to packet loss. The

interarrival jitter field is only a snapshot of the jitter at the

time of a report and is not intended to be taken quantitatively.

Rather, it is intended for comparison across a number of reports from

one receiver over time or from multiple receivers, e.g., within a

single network, at the same time. To allow comparison across

receivers, it is important the the jitter be calculated according to

the same formula by all receivers.

Because the jitter calculation is based on the RTP timestamp which

represents the instant when the first data in the packet was sampled,

any variation in the delay between that sampling instant and the time

the packet is transmitted will affect the resulting jitter that is

calculated. Such a variation in delay would occur for audio packets

of varying duration. It will also occur for video encodings because

the timestamp is the same for all the packets of one frame but those

packets are not all transmitted at the same time. The variation in

delay until transmission does reduce the accuracy of the jitter

calculation as a measure of the behavior of the network by itself,

but it is appropriate to include considering that the receiver buffer

must accommodate it. When the jitter calculation is used as a

comparative measure, the (constant) component due to variation in

delay until transmission subtracts out so that a change in the

Schulzrinne, et al. Standards Track [Page 44]

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network jitter component can then be observed unless it is relatively

small. If the change is small, then it is likely to be

inconsequential.

6.5 SDES: Source Description RTCP Packet

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

header |V=2|P| SC | PT=SDES=202 | length |

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

chunk | SSRC/CSRC_1 |

1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| SDES items |

| ... |

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

chunk | SSRC/CSRC_2 |

2 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| SDES items |

| ... |

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

The SDES packet is a three-level structure composed of a header and

zero or more chunks, each of which is composed of items describing

the source identified in that chunk. The items are described

individually in subsequent sections.

version (V), padding (P), length:

As described for the SR packet (see Section 6.4.1).

packet type (PT): 8 bits

Contains the constant 202 to identify this as an RTCP SDES packet.

source count (SC): 5 bits

The number of SSRC/CSRC chunks contained in this SDES packet. A

value of zero is valid but useless.

Each chunk consists of an SSRC/CSRC identifier followed by a list of

zero or more items, which carry information about the SSRC/CSRC.

Each chunk starts on a 32-bit boundary. Each item consists of an 8-

bit type field, an 8-bit octet count describing the length of the

text (thus, not including this two-octet header), and the text

itself. Note that the text can be no longer than 255 octets, but

this is consistent with the need to limit RTCP bandwidth consumption.

Schulzrinne, et al. Standards Track [Page 45]

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The text is encoded according to the UTF-8 encoding specified in RFC

2279 [5]. US-ASCII is a subset of this encoding and requires no

additional encoding. The presence of multi-octet encodings is

indicated by setting the most significant bit of a character to a

value of one.

Items are contiguous, i.e., items are not individually padded to a

32-bit boundary. Text is not null terminated because some multioctet

encodings include null octets. The list of items in each chunk

MUST be terminated by one or more null octets, the first of which is

interpreted as an item type of zero to denote the end of the list.

No length octet follows the null item type octet, but additional null

octets MUST be included if needed to pad until the next 32-bit

boundary. Note that this padding is separate from that indicated by

the P bit in the RTCP header. A chunk with zero items (four null

octets) is valid but useless.

End systems send one SDES packet containing their own source

identifier (the same as the SSRC in the fixed RTP header). A mixer

sends one SDES packet containing a chunk for each contributing source

from which it is receiving SDES information, or multiple complete

SDES packets in the format above if there are more than 31 such

sources (see Section 7).

The SDES items currently defined are described in the next sections.

Only the CNAME item is mandatory. Some items shown here may be

useful only for particular profiles, but the item types are all

assigned from one common space to promote shared use and to simplify

profile-independent applications. Additional items may be defined in

a profile by registering the type numbers with IANA as described in

Section 15.

6.5.1 CNAME: Canonical End-Point Identifier SDES Item

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| CNAME=1 | length | user and domain name ...

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The CNAME identifier has the following properties:

o Because the randomly allocated SSRC identifier may change if a

conflict is discovered or if a program is restarted, the CNAME

item MUST be included to provide the binding from the SSRC

identifier to an identifier for the source (sender or receiver)

that remains constant.

Schulzrinne, et al. Standards Track [Page 46]

RFC 3550 RTP July 2003

o Like the SSRC identifier, the CNAME identifier SHOULD also be

unique among all participants within one RTP session.

o To provide a binding across multiple media tools used by one

participant in a set of related RTP sessions, the CNAME SHOULD be

fixed for that participant.

o To facilitate third-party monitoring, the CNAME SHOULD be suitable

for either a program or a person to locate the source.

Therefore, the CNAME SHOULD be derived algorithmically and not

entered manually, when possible. To meet these requirements, the

following format SHOULD be used unless a profile specifies an

alternate syntax or semantics. The CNAME item SHOULD have the format

"user@host", or "host" if a user name is not available as on singleuser

systems. For both formats, "host" is either the fully qualified

domain name of the host from which the real-time data originates,

formatted according to the rules specified in RFC 1034 [6], RFC 1035

[7] and Section 2.1 of RFC 1123 [8]; or the standard ASCII

representation of the host's numeric address on the interface used

for the RTP communication. For example, the standard ASCII

representation of an IP Version 4 address is "dotted decimal", also

known as dotted quad, and for IP Version 6, addresses are textually

represented as groups of hexadecimal digits separated by colons (with

variations as detailed in RFC 3513 [23]). Other address types are

expected to have ASCII representations that are mutually unique. The

fully qualified domain name is more convenient for a human observer

and may avoid the need to send a NAME item in addition, but it may be

difficult or impossible to obtain reliably in some operating

environments. Applications that may be run in such environments

SHOULD use the ASCII representation of the address instead.

Examples are "[e-mail]", "doe@192.0.2.89" or

"doe@2201:056D::112E:144A:1E24" for a multi-user system. On a system

with no user name, examples would be "sleepy.example.com",

"192.0.2.89" or "2201:056D::112E:144A:1E24".

The user name SHOULD be in a form that a program such as "finger" or

"talk" could use, i.e., it typically is the login name rather than

the personal name. The host name is not necessarily identical to the

one in the participant's electronic mail address.

This syntax will not provide unique identifiers for each source if an

application permits a user to generate multiple sources from one

host. Such an application would have to rely on the SSRC to further

identify the source, or the profile for that application would have

to specify additional syntax for the CNAME identifier.

Schulzrinne, et al. Standards Track [Page 47]

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If each application creates its CNAME independently, the resulting

CNAMEs may not be identical as would be required to provide a binding

across multiple media tools belonging to one participant in a set of

related RTP sessions. If cross-media binding is required, it may be

necessary for the CNAME of each tool to be externally configured with

the same value by a coordination tool.

Application writers should be aware that private network address

assignments such as the Net-10 assignment proposed in RFC 1918 [24]

may create network addresses that are not globally unique. This

would lead to non-unique CNAMEs if hosts with private addresses and

no direct IP connectivity to the public Internet have their RTP

packets forwarded to the public Internet through an RTP-level

translator. (See also RFC 1627 [25].) To handle this case,

applications MAY provide a means to configure a unique CNAME, but the

burden is on the translator to translate CNAMEs from private

addresses to public addresses if necessary to keep private addresses

from being exposed.

6.5.2 NAME: User Name SDES Item

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| NAME=2 | length | common name of source ...

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

This is the real name used to describe the source, e.g., "John Doe,

Bit Recycler". It may be in any form desired by the user. For

applications such as conferencing, this form of name may be the most

desirable for display in participant lists, and therefore might be

sent most frequently of those items other than CNAME. Profiles MAY

establish such priorities. The NAME value is expected to remain

constant at least for the duration of a session. It SHOULD NOT be

relied upon to be unique among all participants in the session.

6.5.3 EMAIL: Electronic Mail Address SDES Item

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| EMAIL=3 | length | email address of source ...

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The email address is formatted according to RFC 2822 [9], for

example, "[e-mail]". The EMAIL value is expected to

remain constant for the duration of a session.

Schulzrinne, et al. Standards Track [Page 48]

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6.5.4 PHONE: Phone Number SDES Item

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| PHONE=4 | length | phone number of source ...

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The phone number SHOULD be formatted with the plus sign replacing the

international access code. For example, "[phone]" for a

number in the United States.

6.5.5 LOC: Geographic User Location SDES Item

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| LOC=5 | length | geographic location of site ...

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Depending on the application, different degrees of detail are

appropriate for this item. For conference applications, a string

like "Murray Hill, New Jersey" may be sufficient, while, for an

active badge system, strings like "Room 2A244, AT&T BL MH" might be

appropriate. The degree of detail is left to the implementation

and/or user, but format and content MAY be prescribed by a profile.

The LOC value is expected to remain constant for the duration of a

session, except for mobile hosts.

6.5.6 TOOL: Application or Tool Name SDES Item

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| TOOL=6 | length |name/version of source appl. ...

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

A string giving the name and possibly version of the application

generating the stream, e.g., "videotool 1.2". This information may

be useful for debugging purposes and is similar to the Mailer or

Mail-System-Version SMTP headers. The TOOL value is expected to

remain constant for the duration of the session.

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6.5.7 NOTE: Notice/Status SDES Item

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| NOTE=7 | length | note about the source ...

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The following semantics are suggested for this item, but these or

other semantics MAY be explicitly defined by a profile. The NOTE

item is intended for transient messages describing the current state

of the source, e.g., "on the phone, can't talk". Or, during a

seminar, this item might be used to convey the title of the talk. It

should be used only to carry exceptional information and SHOULD NOT

be included routinely by all participants because this would slow

down the rate at which reception reports and CNAME are sent, thus

impairing the performance of the protocol. In particular, it SHOULD

NOT be included as an item in a user's configuration file nor

automatically generated as in a quote-of-the-day.

Since the NOTE item may be important to display while it is active,

the rate at which other non-CNAME items such as NAME are transmitted

might be reduced so that the NOTE item can take that part of the RTCP

bandwidth. When the transient message becomes inactive, the NOTE

item SHOULD continue to be transmitted a few times at the same

repetition rate but with a string of length zero to signal the

receivers. However, receivers SHOULD also consider the NOTE item

inactive if it is not received for a small multiple of the repetition

rate, or perhaps 20-30 RTCP intervals.

6.5.8 PRIV: Private Extensions SDES Item

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| PRIV=8 | length | prefix length |prefix string...

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

... | value string ...

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

This item is used to define experimental or application-specific SDES

extensions. The item contains a prefix consisting of a length-string

pair, followed by the value string filling the remainder of the item

and carrying the desired information. The prefix length field is 8

bits long. The prefix string is a name chosen by the person defining

the PRIV item to be unique with respect to other PRIV items this

application might receive. The application creator might choose to

use the application name plus an additional subtype identification if

Schulzrinne, et al. Standards Track [Page 50]

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needed. Alternatively, it is RECOMMENDED that others choose a name

based on the entity they represent, then coordinate the use of the

name within that entity.

Note that the prefix consumes some space within the item's total

length of 255 octets, so the prefix should be kept as short as

possible. This facility and the constrained RTCP bandwidth SHOULD

NOT be overloaded; it is not intended to satisfy all the control

communication requirements of all applications.

SDES PRIV prefixes will not be registered by IANA. If some form of

the PRIV item proves to be of general utility, it SHOULD instead be

assigned a regular SDES item type registered with IANA so that no

prefix is required. This simplifies use and increases transmission

efficiency.

6.6 BYE: Goodbye RTCP Packet

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

|V=2|P| SC | PT=BYE=203 | length |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| SSRC/CSRC |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

: ... :

+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

(opt) | length | reason for leaving ...

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The BYE packet indicates that one or more sources are no longer

active.

version (V), padding (P), length:

As described for the SR packet (see Section 6.4.1).

packet type (PT): 8 bits

Contains the constant 203 to identify this as an RTCP BYE packet.

source count (SC): 5 bits

The number of SSRC/CSRC identifiers included in this BYE packet.

A count value of zero is valid, but useless.

The rules for when a BYE packet should be sent are specified in

Sections 6.3.7 and 8.2.

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If a BYE packet is received by a mixer, the mixer SHOULD forward the

BYE packet with the SSRC/CSRC identifier(s) unchanged. If a mixer

shuts down, it SHOULD send a BYE packet listing all contributing

sources it handles, as well as its own SSRC identifier. Optionally,

the BYE packet MAY include an 8-bit octet count followed by that many

octets of text indicating the reason for leaving, e.g., "camera

malfunction" or "RTP loop detected". The string has the same

encoding as that described for SDES. If the string fills the packet

to the next 32-bit boundary, the string is not null terminated. If

not, the BYE packet MUST be padded with null octets to the next 32-

bit boundary. This padding is separate from that indicated by the P

bit in the RTCP header.

6.7 APP: Application-Defined RTCP Packet

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

|V=2|P| subtype | PT=APP=204 | length |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| SSRC/CSRC |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| name (ASCII) |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

| application-dependent data ...

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The APP packet is intended for experimental use as new applications

and new features are developed, without requiring packet type value

registration. APP packets with unrecognized names SHOULD be ignored.

After testing and if wider use is justified, it is RECOMMENDED that

each APP packet be redefined without the subtype and name fields and

registered with IANA using an RTCP packet type.

version (V), padding (P), length:

As described for the SR packet (see Section 6.4.1).

subtype: 5 bits

May be used as a subtype to allow a set of APP packets to be

defined under one unique name, or for any application-dependent

data.

packet type (PT): 8 bits

Contains the constant 204 to identify this as an RTCP APP packet.

Schulzrinne, et al. Standards Track [Page 52]

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name: 4 octets

A name chosen by the person defining the set of APP packets to be

unique with respect to other APP packets this application might

receive. The application creator might choose to use the

application name, and then coordinate the allocation of subtype

values to others who want to define new packet types for the

application. Alternatively, it is RECOMMENDED that others choose

a name based on the entity they represent, then coordinate the use

of the name within that entity. The name is interpreted as a

sequence of four ASCII characters, with uppercase and lowercase

characters treated as distinct.

application-dependent data: variable length

Application-dependent data may or may not appear in an APP packet.

It is interpreted by the application and not RTP itself. It MUST

be a multiple of 32 bits long.

7. RTP Translators and Mixers

In addition to end systems, RTP supports the notion of "translators"

and "mixers", which could be considered as "intermediate systems" at

the RTP level. Although this support adds some complexity to the

protocol, the need for these functions has been clearly established

by experiments with multicast audio and video applications in the

Internet. Example uses of translators and mixers given in Section

2.3 stem from the presence of firewalls and low bandwidth

connections, both of which are likely to remain.

7.1 General Description

An RTP translator/mixer connects two or more transport-level

"clouds". Typically, each cloud is defined by a common network and

transport protocol (e.g., IP/UDP) plus a multicast address and

transport level destination port or a pair of unicast addresses and

ports. (Network-level protocol translators, such as IP version 4 to

IP version 6, may be present within a cloud invisibly to RTP.) One

system may serve as a translator or mixer for a number of RTP

sessions, but each is considered a logically separate entity.

In order to avoid creating a loop when a translator or mixer is

installed, the following rules MUST be observed:

o Each of the clouds connected by translators and mixers

participating in one RTP session either MUST be distinct from all

the others in at least one of these parameters (protocol, address,

port), or MUST be isolated at the network level from the others.

Schulzrinne, et al. Standards Track [Page 53]

RFC 3550 RTP July 2003

o A derivative of the first rule is that there MUST NOT be multiple

translators or mixers connected in parallel unless by some

arrangement they partition the set of sources to be forwarded.

Similarly, all RTP end systems that can communicate through one or

more RTP translators or mixers share the same SSRC space, that is,

the SSRC identifiers MUST be unique among all these end systems.

Section 8.2 describes the collision resolution algorithm by which

SSRC identifiers are kept unique and loops are detected.

There may be many varieties of translators and mixers designed for

different purposes and applications. Some examples are to add or

remove encryption, change the encoding of the data or the underlying

protocols, or replicate between a multicast address and one or more

unicast addresses. The distinction between translators and mixers is

that a translator passes through the data streams from different

sources separately, whereas a mixer combines them to form one new

stream:

Translator: Forwards RTP packets with their SSRC identifier

intact; this makes it possible for receivers to identify

individual sources even though packets from all the sources pass

through the same translator and carry the translator's network

source address. Some kinds of translators will pass through the

data untouched, but others MAY change the encoding of the data and

thus the RTP data payload type and timestamp. If multiple data

packets are re-encoded into one, or vice versa, a translator MUST

assign new sequence numbers to the outgoing packets. Losses in

the incoming packet stream may induce corresponding gaps in the

outgoing sequence numbers. Receivers cannot detect the presence

of a translator unless they know by some other means what payload

type or transport address was used by the original source.

Mixer: Receives streams of RTP data packets from one or more

sources, possibly changes the data format, combines the streams in

some manner and then forwards the combined stream. Since the

timing among multiple input sources will not generally be

synchronized, the mixer will make timing adjustments among the

streams and generate its own timing for the combined stream, so it

is the synchronization source. Thus, all data packets forwarded

by a mixer MUST be marked with the mixer's own SSRC identifier.

In order to preserve the identity of the original sources

contributing to the mixed packet, the mixer SHOULD insert their

SSRC identifiers into the CSRC identifier list following the fixed

RTP header of the packet. A mixer that is also itself a

contributing source for some packet SHOULD explicitly include its

own SSRC identifier in the CSRC list for that packet.

Schulzrinne, et al. Standards Track [Page 54]

RFC 3550 RTP July 2003

For some applications, it MAY be acceptable for a mixer not to

identify sources in the CSRC list. However, this introduces the

danger that loops involving those sources could not be detected.

The advantage of a mixer over a translator for applications like

audio is that the output bandwidth is limited to that of one source

even when multiple sources are active on the input side. This may be

important for low-bandwidth links. The disadvantage is that

receivers on the output side don't have any control over which

sources are passed through or muted, unless some mechanism is

implemented for remote control of the mixer. The regeneration of

synchronization information by mixers also means that receivers can't

do inter-media synchronization of the original streams. A multimedia

mixer could do it.

[E1] [E6]

| |

E1:17 | E6:15 |

| | E6:15

V M1:48 (1,17) M1:48 (1,17) V M1:48 (1,17)

(M1)-------------><T1>-----------------><T2>-------------->[E7]

^ ^ E4:47 ^ E4:47

E2:1 | E4:47 | | M3:89 (64,45)

| | |

[E2] [E4] M3:89 (64,45) |

| legend:

[E3] --------->(M2)----------->(M3)------------| [End system]

E3:64 M2:12 (64) ^ (Mixer)

| E5:45 <Translator>

|

[E5] source: SSRC (CSRCs)

------------------->

Figure 3: Sample RTP network with end systems, mixers and translators

A collection of mixers and translators is shown in Fig. 3 to

illustrate their effect on SSRC and CSRC identifiers. In the figure,

end systems are shown as rectangles (named E), translators as

triangles (named T) and mixers as ovals (named M). The notation "M1:

48(1,17)" designates a packet originating a mixer M1, identified by

M1's (random) SSRC value of 48 and two CSRC identifiers, 1 and 17,

copied from the SSRC identifiers of packets from E1 and E2.

7.2 RTCP Processing in Translators

In addition to forwarding data packets, perhaps modified, translators

and mixers MUST also process RTCP packets. In many cases, they will

take apart the compound RTCP packets received from end systems to

Schulzrinne, et al. Standards Track [Page 55]

RFC 3550 RTP July 2003

aggregate SDES information and to modify the SR or RR packets.

Retransmission of this information may be triggered by the packet

arrival or by the RTCP interval timer of the translator or mixer

itself.

A translator that does not modify the data packets, for example one

that just replicates between a multicast address and a unicast

address, MAY simply forward RTCP packets unmodified as well. A

translator that transforms the payload in some way MUST make

corresponding transformations in the SR and RR information so that it

still reflects the characteristics of the data and the reception

quality. These translators MUST NOT simply forward RTCP packets. In

general, a translator SHOULD NOT aggregate SR and RR packets from

different sources into one packet since that would reduce the

accuracy of the propagation delay measurements based on the LSR and

DLSR fields.

SR sender information: A translator does not generate its own

sender information, but forwards the SR packets received from one

cloud to the others. The SSRC is left intact but the sender

information MUST be modified if required by the translation. If a

translator changes the data encoding, it MUST change the "sender's

byte count" field. If it also combines several data packets into

one output packet, it MUST change the "sender's packet count"

field. If it changes the timestamp frequency, it MUST change the

"RTP timestamp" field in the SR packet.

SR/RR reception report blocks: A translator forwards reception

reports received from one cloud to the others. Note that these

flow in the direction opposite to the data. The SSRC is left

intact. If a translator combines several data packets into one

output packet, and therefore changes the sequence numbers, it MUST

make the inverse manipulation for the packet loss fields and the

"extended last sequence number" field. This may be complex. In

the extreme case, there may be no meaningful way to translate the

reception reports, so the translator MAY pass on no reception

report at all or a synthetic report based on its own reception.

The general rule is to do what makes sense for a particular

translation.

A translator does not require an SSRC identifier of its own, but

MAY choose to allocate one for the purpose of sending reports

about what it has received. These would be sent to all the

connected clouds, each corresponding to the translation of the

data stream as sent to that cloud, since reception reports are

normally multicast to all participants.

Schulzrinne, et al. Standards Track [Page 56]

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SDES: Translators typically forward without change the SDES

information they receive from one cloud to the others, but MAY,

for example, decide to filter non-CNAME SDES information if

bandwidth is limited. The CNAMEs MUST be forwarded to allow SSRC

identifier collision detection to work. A translator that

generates its own RR packets MUST send SDES CNAME information

about itself to the same clouds that it sends those RR packets.

BYE: Translators forward BYE packets unchanged. A translator

that is about to cease forwarding packets SHOULD send a BYE packet

to each connected cloud containing all the SSRC identifiers that

were previously being forwarded to that cloud, including the

translator's own SSRC identifier if it sent reports of its own.

APP: Translators forward APP packets unchanged.

7.3 RTCP Processing in Mixers

Since a mixer generates a new data stream of its own, it does not

pass through SR or RR packets at all and instead generates new

information for both sides.

SR sender information: A mixer does not pass through sender

information from the sources it mixes because the characteristics

of the source streams are lost in the mix. As a synchronization

source, the mixer SHOULD generate its own SR packets with sender

information about the mixed data stream and send them in the same

direction as the mixed stream.

SR/RR reception report blocks: A mixer generates its own

reception reports for sources in each cloud and sends them out

only to the same cloud. It MUST NOT send these reception reports

to the other clouds and MUST NOT forward reception reports from

one cloud to the others because the sources would not be SSRCs

there (only CSRCs).

SDES: Mixers typically forward without change the SDES

information they receive from one cloud to the others, but MAY,

for example, decide to filter non-CNAME SDES information if

bandwidth is limited. The CNAMEs MUST be forwarded to allow SSRC

identifier collision detection to work. (An identifier in a CSRC

list generated by a mixer might collide with an SSRC identifier

generated by an end system.) A mixer MUST send SDES CNAME

information about itself to the same clouds that it sends SR or RR

packets.

Schulzrinne, et al. Standards Track [Page 57]

RFC 3550 RTP July 2003

Since mixers do not forward SR or RR packets, they will typically

be extracting SDES packets from a compound RTCP packet. To

minimize overhead, chunks from the SDES packets MAY be aggregated

into a single SDES packet which is then stacked on an SR or RR

packet originating from the mixer. A mixer which aggregates SDES

packets will use more RTCP bandwidth than an individual source

because the compound packets will be longer, but that is

appropriate since the mixer represents multiple sources.

Similarly, a mixer which passes through SDES packets as they are

received will be transmitting RTCP packets at higher than the

single source rate, but again that is correct since the packets

come from multiple sources. The RTCP packet rate may be different

on each side of the mixer.

A mixer that does not insert CSRC identifiers MAY also refrain

from forwarding SDES CNAMEs. In this case, the SSRC identifier

spaces in the two clouds are independent. As mentioned earlier,

this mode of operation creates a danger that loops can't be

detected.

BYE: Mixers MUST forward BYE packets. A mixer that is about to

cease forwarding packets SHOULD send a BYE packet to each

connected cloud containing all the SSRC identifiers that were

previously being forwarded to that cloud, including the mixer's

own SSRC identifier if it sent reports of its own.

APP: The treatment of APP packets by mixers is application-specific.

7.4 Cascaded Mixers

An RTP session may involve a collection of mixers and translators as

shown in Fig. 3. If two mixers are cascaded, such as M2 and M3 in

the figure, packets received by a mixer may already have been mixed

and may include a CSRC list with multiple identifiers. The second

mixer SHOULD build the CSRC list for the outgoing packet using the

CSRC identifiers from already-mixed input packets and the SSRC

identifiers from unmixed input packets. This is shown in the output

arc from mixer M3 labeled M3:89(64,45) in the figure. As in the case

of mixers that are not cascaded, if the resulting CSRC list has more

than 15 identifiers, the remainder cannot be included.

Schulzrinne, et al. Standards Track [Page 58]

RFC 3550 RTP July 2003

8. SSRC Identifier Allocation and Use

The SSRC identifier carried in the RTP header and in various fields

of RTCP packets is a random 32-bit number that is required to be

globally unique within an RTP session. It is crucial that the number

be chosen with care in order that participants on the same network or

starting at the same time are not likely to choose the same number.

It is not sufficient to use the local network address (such as an

IPv4 address) for the identifier because the address may not be

unique. Since RTP translators and mixers enable interoperation among

multiple networks with different address spaces, the allocation

patterns for addresses within two spaces might result in a much

higher rate of collision than would occur with random allocation.

Multiple sources running on one host would also conflict.

It is also not sufficient to obtain an SSRC identifier simply by

calling random() without carefully initializing the state. An

example of how to generate a random identifier is presented in

Appendix A.6.

8.1 Probability of Collision

Since the identifiers are chosen randomly, it is possible that two or

more sources will choose the same number. Collision occurs with the

highest probability when all sources are started simultaneously, for

example when triggered automatically by some session management

event. If N is the number of sources and L the length of the

identifier (here, 32 bits), the probability that two sources

independently pick the same value can be approximated for large N

[26] as 1 - exp(-N**2 / 2**(L+1)). For N=1000, the probability is

roughly 10**-4.

The typical collision probability is much lower than the worst-case

above. When one new source joins an RTP session in which all the

other sources already have unique identifiers, the probability of

collision is just the fraction of numbers used out of the space.

Again, if N is the number of sources and L the length of the

identifier, the probability of collision is N / 2**L. For N=1000,

the probability is roughly 2*10**-7.

The probability of collision is further reduced by the opportunity

for a new source to receive packets from other participants before

sending its first packet (either data or control). If the new source

keeps track of the other participants (by SSRC identifier), then

Schulzrinne, et al. Standards Track [Page 59]

RFC 3550 RTP July 2003

before transmitting its first packet the new source can verify that

its identifier does not conflict with any that have been received, or

else choose again.

8.2 Collision Resolution and Loop Detection

Although the probability of SSRC identifier collision is low, all RTP

implementations MUST be prepared to detect collisions and take the

appropriate actions to resolve them. If a source discovers at any

time that another source is using the same SSRC identifier as its

own, it MUST send an RTCP BYE packet for the old identifier and

choose another random one. (As explained below, this step is taken

only once in case of a loop.) If a receiver discovers that two other

sources are colliding, it MAY keep the packets from one and discard

the packets from the other when this can be detected by different

source transport addresses or CNAMEs. The two sources are expected

to resolve the collision so that the situation doesn't last.

Because the random SSRC identifiers are kept globally unique for each

RTP session, they can also be used to detect loops that may be

introduced by mixers or translators. A loop causes duplication of

data and control information, either unmodified or possibly mixed, as

in the following examples:

o A translator may incorrectly forward a packet to the same

multicast group from which it has received the packet, either

directly or through a chain of translators. In that case, the

same packet appears several times, originating from different

network sources.

o Two translators incorrectly set up in parallel, i.e., with the

same multicast groups on both sides, would both forward packets

from one multicast group to the other. Unidirectional translators

would produce two copies; bidirectional translators would form a

loop.

o A mixer can close a loop by sending to the same transport

destination upon which it receives packets, either directly or

through another mixer or translator. In this case a source might

show up both as an SSRC on a data packet and a CSRC in a mixed

data packet.

A source may discover that its own packets are being looped, or that

packets from another source are being looped (a third-party loop).

Both loops and collisions in the random selection of a source

identifier result in packets arriving with the same SSRC identifier

but a different source transport address, which may be that of the

end system originating the packet or an intermediate system.

Schulzrinne, et al. Standards Track [Page 60]

RFC 3550 RTP July 2003

Therefore, if a source changes its source transport address, it MAY

also choose a new SSRC identifier to avoid being interpreted as a

looped source. (This is not MUST because in some applications of RTP

sources may be expected to change addresses during a session.) Note

that if a translator restarts and consequently changes the source

transport address (e.g., changes the UDP source port number) on which

it forwards packets, then all those packets will appear to receivers

to be looped because the SSRC identifiers are applied by the original

source and will not change. This problem can be avoided by keeping

the source transport address fixed across restarts, but in any case

will be resolved after a timeout at the receivers.

Loops or collisions occurring on the far side of a translator or

mixer cannot be detected using the source transport address if all

copies of the packets go through the translator or mixer, however,

collisions may still be detected when chunks from two RTCP SDES

packets contain the same SSRC identifier but different CNAMEs.

To detect and resolve these conflicts, an RTP implementation MUST

include an algorithm similar to the one described below, though the

implementation MAY choose a different policy for which packets from

colliding third-party sources are kept. The algorithm described

below ignores packets from a new source or loop that collide with an

established source. It resolves collisions with the participant's

own SSRC identifier by sending an RTCP BYE for the old identifier and

choosing a new one. However, when the collision was induced by a

loop of the participant's own packets, the algorithm will choose a

new identifier only once and thereafter ignore packets from the

looping source transport address. This is required to avoid a flood

of BYE packets.

This algorithm requires keeping a table indexed by the source

identifier and containing the source transport addresses from the

first RTP packet and first RTCP packet received with that identifier,

along with other state for that source. Two source transport

addresses are required since, for example, the UDP source port

numbers may be different on RTP and RTCP packets. However, it may be

assumed that the network address is the same in both source transport

addresses.

Each SSRC or CSRC identifier received in an RTP or RTCP packet is

looked up in the source identifier table in order to process that

data or control information. The source transport address from the

packet is compared to the corresponding source transport address in

the table to detect a loop or collision if they don't match. For

control packets, each element with its own SSRC identifier, for

example an SDES chunk, requires a separate lookup. (The SSRC

identifier in a reception report block is an exception because it

Schulzrinne, et al. Standards Track [Page 61]

RFC 3550 RTP July 2003

identifies a source heard by the reporter, and that SSRC identifier

is unrelated to the source transport address of the RTCP packet sent

by the reporter.) If the SSRC or CSRC is not found, a new entry is

created. These table entries are removed when an RTCP BYE packet is

received with the corresponding SSRC identifier and validated by a

matching source transport address, or after no packets have arrived

for a relatively long time (see Section 6.2.1).

Note that if two sources on the same host are transmitting with the

same source identifier at the time a receiver begins operation, it

would be possible that the first RTP packet received came from one of

the sources while the first RTCP packet received came from the other.

This would cause the wrong RTCP information to be associated with the

RTP data, but this situation should be sufficiently rare and harmless

that it may be disregarded.

In order to track loops of the participant's own data packets, the

implementation MUST also keep a separate list of source transport

addresses (not identifiers) that have been found to be conflicting.

As in the source identifier table, two source transport addresses

MUST be kept to separately track conflicting RTP and RTCP packets.

Note that the conflicting address list should be short, usually

empty. Each element in this list stores the source addresses plus

the time when the most recent conflicting packet was received. An

element MAY be removed from the list when no conflicting packet has

arrived from that source for a time on the order of 10 RTCP report

intervals (see Section 6.2).

For the algorithm as shown, it is assumed that the participant's own

source identifier and state are included in the source identifier

table. The algorithm could be restructured to first make a separate

comparison against the participant's own source identifier.

if (SSRC or CSRC identifier is not found in the source

identifier table) {

create a new entry storing the data or control source

transport address, the SSRC or CSRC and other state;

}

/* Identifier is found in the table */

else if (table entry was created on receipt of a control packet

and this is the first data packet or vice versa) {

store the source transport address from this packet;

}

else if (source transport address from the packet does not match

the one saved in the table entry for this identifier) {

Schulzrinne, et al. Standards Track [Page 62]

RFC 3550 RTP July 2003

/* An identifier collision or a loop is indicated */

if (source identifier is not the participant's own) {

/* OPTIONAL error counter step */

if (source identifier is from an RTCP SDES chunk

containing a CNAME item that differs from the CNAME

in the table entry) {

count a third-party collision;

} else {

count a third-party loop;

}

abort processing of data packet or control element;

/* MAY choose a different policy to keep new source */

}

/* A collision or loop of the participant's own packets */

else if (source transport address is found in the list of

conflicting data or control source transport

addresses) {

/* OPTIONAL error counter step */

if (source identifier is not from an RTCP SDES chunk

containing a CNAME item or CNAME is the

participant's own) {

count occurrence of own traffic looped;

}

mark current time in conflicting address list entry;

abort processing of data packet or control element;

}

/* New collision, change SSRC identifier */

else {

log occurrence of a collision;

create a new entry in the conflicting data or control

source transport address list and mark current time;

send an RTCP BYE packet with the old SSRC identifier;

choose a new SSRC identifier;

create a new entry in the source identifier table with

the old SSRC plus the source transport address from

the data or control packet being processed;

}

}

In this algorithm, packets from a newly conflicting source address

will be ignored and packets from the original source address will be

kept. If no packets arrive from the original source for an extended

period, the table entry will be timed out and the new source will be

Schulzrinne, et al. Standards Track [Page 63]

RFC 3550 RTP July 2003

able to take over. This might occur if the original source detects

the collision and moves to a new source identifier, but in the usual

case an RTCP BYE packet will be received from the original source to

delete the state without having to wait for a timeout.

If the original source address was received through a mixer (i.e.,

learned as a CSRC) and later the same source is received directly,

the receiver may be well advised to switch to the new source address

unless other sources in the mix would be lost. Furthermore, for

applications such as telephony in which some sources such as mobile

entities may change addresses during the course of an RTP session,

the RTP implementation SHOULD modify the collision detection

algorithm to accept packets from the new source transport address.

To guard against flip-flopping between addresses if a genuine

collision does occur, the algorithm SHOULD include some means to

detect this case and avoid switching.

When a new SSRC identifier is chosen due to a collision, the

candidate identifier SHOULD first be looked up in the source

identifier table to see if it was already in use by some other

source. If so, another candidate MUST be generated and the process

repeated.

A loop of data packets to a multicast destination can cause severe

network flooding. All mixers and translators MUST implement a loop

detection algorithm like the one here so that they can break loops.

This should limit the excess traffic to no more than one duplicate

copy of the original traffic, which may allow the session to continue

so that the cause of the loop can be found and fixed. However, in

extreme cases where a mixer or translator does not properly break the

loop and high traffic levels result, it may be necessary for end

systems to cease transmitting data or control packets entirely. This

decision may depend upon the application. An error condition SHOULD

be indicated as appropriate. Transmission MAY be attempted again

periodically after a long, random time (on the order of minutes).

8.3 Use with Layered Encodings

For layered encodings transmitted on separate RTP sessions (see

Section 2.4), a single SSRC identifier space SHOULD be used across

the sessions of all layers and the core (base) layer SHOULD be used

for SSRC identifier allocation and collision resolution. When a

source discovers that it has collided, it transmits an RTCP BYE

packet on only the base layer but changes the SSRC identifier to the

new value in all layers.

Schulzrinne, et al. Standards Track [Page 64]

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9. Security

Lower layer protocols may eventually provide all the security

services that may be desired for applications of RTP, including

authentication, integrity, and confidentiality. These services have

been specified for IP in [27]. Since the initial audio and video

applications using RTP needed a confidentiality service before such

services were available for the IP layer, the confidentiality service

described in the next section was defined for use with RTP and RTCP.

That description is included here to codify existing practice. New

applications of RTP MAY implement this RTP-specific confidentiality

service for backward compatibility, and/or they MAY implement

alternative security services. The overhead on the RTP protocol for

this confidentiality service is low, so the penalty will be minimal

if this service is obsoleted by other services in the future.

Alternatively, other services, other implementations of services and

other algorithms may be defined for RTP in the future. In

particular, an RTP profile called Secure Real-time Transport Protocol

(SRTP) [28] is being developed to provide confidentiality of the RTP

payload while leaving the RTP header in the clear so that link-level

header compression algorithms can still operate. It is expected that

SRTP will be the correct choice for many applications. SRTP is based

on the Advanced Encryption Standard (AES) and provides stronger

security than the service described here. No claim is made that the

methods presented here are appropriate for a particular security

need. A profile may specify which services and algorithms should be

offered by applications, and may provide guidance as to their

appropriate use.

Key distribution and certificates are outside the scope of this

document.

9.1 Confidentiality

Confidentiality means that only the intended receiver(s) can decode

the received packets; for others, the packet contains no useful

information. Confidentiality of the content is achieved by

encryption.

When it is desired to encrypt RTP or RTCP according to the method

specified in this section, all the octets that will be encapsulated

for transmission in a single lower-layer packet are encrypted as a

unit. For RTCP, a 32-bit random number redrawn for each unit MUST be

prepended to the unit before encryption. For RTP, no prefix is

prepended; instead, the sequence number and timestamp fields are

initialized with random offsets. This is considered to be a weak

Schulzrinne, et al. Standards Track [Page 65]

RFC 3550 RTP July 2003

initialization vector (IV) because of poor randomness properties. In

addition, if the subsequent field, the SSRC, can be manipulated by an

enemy, there is further weakness of the encryption method.

For RTCP, an implementation MAY segregate the individual RTCP packets

in a compound RTCP packet into two separate compound RTCP packets,

one to be encrypted and one to be sent in the clear. For example,

SDES information might be encrypted while reception reports were sent

in the clear to accommodate third-party monitors that are not privy

to the encryption key. In this example, depicted in Fig. 4, the SDES

information MUST be appended to an RR packet with no reports (and the

random number) to satisfy the requirement that all compound RTCP

packets begin with an SR or RR packet. The SDES CNAME item is

required in either the encrypted or unencrypted packet, but not both.

The same SDES information SHOULD NOT be carried in both packets as

this may compromise the encryption.

UDP packet UDP packet

----------------------------- ------------------------------

[random][RR][SDES #CNAME ...] [SR #senderinfo #site1 #site2]

----------------------------- ------------------------------

encrypted not encrypted

#: SSRC identifier

Figure 4: Encrypted and non-encrypted RTCP packets

The presence of encryption and the use of the correct key are

confirmed by the receiver through header or payload validity checks.

Examples of such validity checks for RTP and RTCP headers are given

in Appendices A.1 and A.2.

To be consistent with existing implementations of the initial

specification of RTP in RFC 1889, the default encryption algorithm is

the Data Encryption Standard (DES) algorithm in cipher block chaining

(CBC) mode, as described in Section 1.1 of RFC 1423 [29], except that

padding to a multiple of 8 octets is indicated as described for the P

bit in Section 5.1. The initialization vector is zero because random

values are supplied in the RTP header or by the random prefix for

compound RTCP packets. For details on the use of CBC initialization

vectors, see [30].

Implementations that support the encryption method specified here

SHOULD always support the DES algorithm in CBC mode as the default

cipher for this method to maximize interoperability. This method was

chosen because it has been demonstrated to be easy and practical to

use in experimental audio and video tools in operation on the

Internet. However, DES has since been found to be too easily broken.

Schulzrinne, et al. Standards Track [Page 66]

RFC 3550 RTP July 2003

It is RECOMMENDED that stronger encryption algorithms such as

Triple-DES be used in place of the default algorithm. Furthermore,

secure CBC mode requires that the first block of each packet be XORed

with a random, independent IV of the same size as the cipher's block

size. For RTCP, this is (partially) achieved by prepending each

packet with a 32-bit random number, independently chosen for each

packet. For RTP, the timestamp and sequence number start from random

values, but consecutive packets will not be independently randomized.

It should be noted that the randomness in both cases (RTP and RTCP)

is limited. High-security applications SHOULD consider other, more

conventional, protection means. Other encryption algorithms MAY be

specified dynamically for a session by non-RTP means. In particular,

the SRTP profile [28] based on AES is being developed to take into

account known plaintext and CBC plaintext manipulation concerns, and

will be the correct choice in the future.

As an alternative to encryption at the IP level or at the RTP level

as described above, profiles MAY define additional payload types for

encrypted encodings. Those encodings MUST specify how padding and

other aspects of the encryption are to be handled. This method

allows encrypting only the data while leaving the headers in the

clear for applications where that is desired. It may be particularly

useful for hardware devices that will handle both decryption and

decoding. It is also valuable for applications where link-level

compression of RTP and lower-layer headers is desired and

confidentiality of the payload (but not addresses) is sufficient

since encryption of the headers precludes compression.

9.2 Authentication and Message Integrity

Authentication and message integrity services are not defined at the

RTP level since these services would not be directly feasible without

a key management infrastructure. It is expected that authentication

and integrity services will be provided by lower layer protocols.

10. Congestion Control

All transport protocols used on the Internet need to address

congestion control in some way [31]. RTP is not an exception, but

because the data transported over RTP is often inelastic (generated

at a fixed or controlled rate), the means to control congestion in

RTP may be quite different from those for other transport protocols

such as TCP. In one sense, inelasticity reduces the risk of

congestion because the RTP stream will not expand to consume all

available bandwidth as a TCP stream can. However, inelasticity also

means that the RTP stream cannot arbitrarily reduce its load on the

network to eliminate congestion when it occurs.

Schulzrinne, et al. Standards Track [Page 67]

RFC 3550 RTP July 2003

Since RTP may be used for a wide variety of applications in many

different contexts, there is no single congestion control mechanism

that will work for all. Therefore, congestion control SHOULD be

defined in each RTP profile as appropriate. For some profiles, it

may be sufficient to include an applicability statement restricting

the use of that profile to environments where congestion is avoided

by engineering. For other profiles, specific methods such as data

rate adaptation based on RTCP feedback may be required.

11. RTP over Network and Transport Protocols

This section describes issues specific to carrying RTP packets within

particular network and transport protocols. The following rules

apply unless superseded by protocol-specific definitions outside this

specification.

RTP relies on the underlying protocol(s) to provide demultiplexing of

RTP data and RTCP control streams. For UDP and similar protocols,

RTP SHOULD use an even destination port number and the corresponding

RTCP stream SHOULD use the next higher (odd) destination port number.

For applications that take a single port number as a parameter and

derive the RTP and RTCP port pair from that number, if an odd number

is supplied then the application SHOULD replace that number with the

next lower (even) number to use as the base of the port pair. For

applications in which the RTP and RTCP destination port numbers are

specified via explicit, separate parameters (using a signaling

protocol or other means), the application MAY disregard the

restrictions that the port numbers be even/odd and consecutive

although the use of an even/odd port pair is still encouraged. The

RTP and RTCP port numbers MUST NOT be the same since RTP relies on

the port numbers to demultiplex the RTP data and RTCP control

streams.

In a unicast session, both participants need to identify a port pair

for receiving RTP and RTCP packets. Both participants MAY use the

same port pair. A participant MUST NOT assume that the source port

of the incoming RTP or RTCP packet can be used as the destination

port for outgoing RTP or RTCP packets. When RTP data packets are

being sent in both directions, each participant's RTCP SR packets

MUST be sent to the port that the other participant has specified for

reception of RTCP. The RTCP SR packets combine sender information

for the outgoing data plus reception report information for the

incoming data. If a side is not actively sending data (see Section

6.4), an RTCP RR packet is sent instead.

It is RECOMMENDED that layered encoding applications (see Section

2.4) use a set of contiguous port numbers. The port numbers MUST be

distinct because of a widespread deficiency in existing operating

Schulzrinne, et al. Standards Track [Page 68]

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systems that prevents use of the same port with multiple multicast

addresses, and for unicast, there is only one permissible address.

Thus for layer n, the data port is P + 2n, and the control port is P

+ 2n + 1. When IP multicast is used, the addresses MUST also be

distinct because multicast routing and group membership are managed

on an address granularity. However, allocation of contiguous IP

multicast addresses cannot be assumed because some groups may require

different scopes and may therefore be allocated from different

address ranges.

The previous paragraph conflicts with the SDP specification, RFC 2327

[15], which says that it is illegal for both multiple addresses and

multiple ports to be specified in the same session description

because the association of addresses with ports could be ambiguous.

It is intended that this restriction will be relaxed in a revision of

RFC 2327 to allow an equal number of addresses and ports to be

specified with a one-to-one mapping implied.

RTP data packets contain no length field or other delineation,

therefore RTP relies on the underlying protocol(s) to provide a

length indication. The maximum length of RTP packets is limited only

by the underlying protocols.

If RTP packets are to be carried in an underlying protocol that

provides the abstraction of a continuous octet stream rather than

messages (packets), an encapsulation of the RTP packets MUST be

defined to provide a framing mechanism. Framing is also needed if

the underlying protocol may contain padding so that the extent of the

RTP payload cannot be determined. The framing mechanism is not

defined here.

A profile MAY specify a framing method to be used even when RTP is

carried in protocols that do provide framing in order to allow

carrying several RTP packets in one lower-layer protocol data unit,

such as a UDP packet. Carrying several RTP packets in one network or

transport packet reduces header overhead and may simplify

synchronization between different streams.

12. Summary of Protocol Constants

This section contains a summary listing of the constants defined in

this specification.

The RTP payload type (PT) constants are defined in profiles rather

than this document. However, the octet of the RTP header which

contains the marker bit(s) and payload type MUST avoid the reserved

values 200 and 201 (decimal) to distinguish RTP packets from the RTCP

SR and RR packet types for the header validation procedure described

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in Appendix A.1. For the standard definition of one marker bit and a

7-bit payload type field as shown in this specification, this

restriction means that payload types 72 and 73 are reserved.

12.1 RTCP Packet Types

abbrev. name value

SR sender report 200

RR receiver report 201

SDES source description 202

BYE goodbye 203

APP application-defined 204

These type values were chosen in the range 200-204 for improved

header validity checking of RTCP packets compared to RTP packets or

other unrelated packets. When the RTCP packet type field is compared

to the corresponding octet of the RTP header, this range corresponds

to the marker bit being 1 (which it usually is not in data packets)

and to the high bit of the standard payload type field being 1 (since

the static payload types are typically defined in the low half).

This range was also chosen to be some distance numerically from 0 and

255 since all-zeros and all-ones are common data patterns.

Since all compound RTCP packets MUST begin with SR or RR, these codes

were chosen as an even/odd pair to allow the RTCP validity check to

test the maximum number of bits with mask and value.

Additional RTCP packet types may be registered through IANA (see

Section 15).

12.2 SDES Types

abbrev. name value

END end of SDES list 0

CNAME canonical name 1

NAME user name 2

EMAIL user's electronic mail address 3

PHONE user's phone number 4

LOC geographic user location 5

TOOL name of application or tool 6

NOTE notice about the source 7

PRIV private extensions 8

Additional SDES types may be registered through IANA (see Section

15).

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13. RTP Profiles and Payload Format Specifications

A complete specification of RTP for a particular application will

require one or more companion documents of two types described here:

profiles, and payload format specifications.

RTP may be used for a variety of applications with somewhat differing

requirements. The flexibility to adapt to those requirements is

provided by allowing multiple choices in the main protocol

specification, then selecting the appropriate choices or defining

extensions for a particular environment and class of applications in

a separate profile document. Typically an application will operate

under only one profile in a particular RTP session, so there is no

explicit indication within the RTP protocol itself as to which

profile is in use. A profile for audio and video applications may be

found in the companion RFC 3551. Profiles are typically titled "RTP

Profile for ...".

The second type of companion document is a payload format

specification, which defines how a particular kind of payload data,

such as H.261 encoded video, should be carried in RTP. These

documents are typically titled "RTP Payload Format for XYZ

Audio/Video Encoding". Payload formats may be useful under multiple

profiles and may therefore be defined independently of any particular

profile. The profile documents are then responsible for assigning a

default mapping of that format to a payload type value if needed.

Within this specification, the following items have been identified

for possible definition within a profile, but this list is not meant

to be exhaustive:

RTP data header: The octet in the RTP data header that contains

the marker bit and payload type field MAY be redefined by a

profile to suit different requirements, for example with more or

fewer marker bits (Section 5.3, p. 18).

Payload types: Assuming that a payload type field is included,

the profile will usually define a set of payload formats (e.g.,

media encodings) and a default static mapping of those formats to

payload type values. Some of the payload formats may be defined

by reference to separate payload format specifications. For each

payload type defined, the profile MUST specify the RTP timestamp

clock rate to be used (Section 5.1, p. 14).

RTP data header additions: Additional fields MAY be appended to

the fixed RTP data header if some additional functionality is

required across the profile's class of applications independent of

payload type (Section 5.3, p. 18).

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RTP data header extensions: The contents of the first 16 bits of

the RTP data header extension structure MUST be defined if use of

that mechanism is to be allowed under the profile for

implementation-specific extensions (Section 5.3.1, p. 18).

RTCP packet types: New application-class-specific RTCP packet

types MAY be defined and registered with IANA.

RTCP report interval: A profile SHOULD specify that the values

suggested in Section 6.2 for the constants employed in the

calculation of the RTCP report interval will be used. Those are

the RTCP fraction of session bandwidth, the minimum report

interval, and the bandwidth split between senders and receivers.

A profile MAY specify alternate values if they have been

demonstrated to work in a scalable manner.

SR/RR extension: An extension section MAY be defined for the

RTCP SR and RR packets if there is additional information that

should be reported regularly about the sender or receivers

(Section 6.4.3, p. 42 and 43).

SDES use: The profile MAY specify the relative priorities for

RTCP SDES items to be transmitted or excluded entirely (Section

6.3.9); an alternate syntax or semantics for the CNAME item

(Section 6.5.1); the format of the LOC item (Section 6.5.5); the

semantics and use of the NOTE item (Section 6.5.7); or new SDES

item types to be registered with IANA.

Security: A profile MAY specify which security services and

algorithms should be offered by applications, and MAY provide

guidance as to their appropriate use (Section 9, p. 65).

String-to-key mapping: A profile MAY specify how a user-provided

password or pass phrase is mapped into an encryption key.

Congestion: A profile SHOULD specify the congestion control

behavior appropriate for that profile.

Underlying protocol: Use of a particular underlying network or

transport layer protocol to carry RTP packets MAY be required.

Transport mapping: A mapping of RTP and RTCP to transport-level

addresses, e.g., UDP ports, other than the standard mapping

defined in Section 11, p. 68 may be specified.

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Encapsulation: An encapsulation of RTP packets may be defined to

allow multiple RTP data packets to be carried in one lower-layer

packet or to provide framing over underlying protocols that do not

already do so (Section 11, p. 69).

It is not expected that a new profile will be required for every

application. Within one application class, it would be better to

extend an existing profile rather than make a new one in order to

facilitate interoperation among the applications since each will

typically run under only one profile. Simple extensions such as the

definition of additional payload type values or RTCP packet types may

be accomplished by registering them through IANA and publishing their

descriptions in an addendum to the profile or in a payload format

specification.

14. Security Considerations

RTP suffers from the same security liabilities as the underlying

protocols. For example, an impostor can fake source or destination

network addresses, or change the header or payload. Within RTCP, the

CNAME and NAME information may be used to impersonate another

participant. In addition, RTP may be sent via IP multicast, which

provides no direct means for a sender to know all the receivers of

the data sent and therefore no measure of privacy. Rightly or not,

users may be more sensitive to privacy concerns with audio and video

communication than they have been with more traditional forms of

network communication [33]. Therefore, the use of security

mechanisms with RTP is important. These mechanisms are discussed in

Section 9.

RTP-level translators or mixers may be used to allow RTP traffic to

reach hosts behind firewalls. Appropriate firewall security

principles and practices, which are beyond the scope of this

document, should be followed in the design and installation of these

devices and in the admission of RTP applications for use behind the

firewall.

15. IANA Considerations

Additional RTCP packet types and SDES item types may be registered

through the Internet Assigned Numbers Authority (IANA). Since these

number spaces are small, allowing unconstrained registration of new

values would not be prudent. To facilitate review of requests and to

promote shared use of new types among multiple applications, requests

for registration of new values must be documented in an RFC or other

permanent and readily available reference such as the product of

another cooperative standards body (e.g., ITU-T). Other requests may

also be accepted, under the advice of a "designated expert."

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(Contact the IANA for the contact information of the current expert.)

RTP profile specifications SHOULD register with IANA a name for the

profile in the form "RTP/xxx", where xxx is a short abbreviation of

the profile title. These names are for use by higher-level control

protocols, such as the Session Description Protocol (SDP), RFC 2327

[15], to refer to transport methods.

16. Intellectual Property Rights Statement

The IETF takes no position regarding the validity or scope of any

intellectual property or other rights that might be claimed to

pertain to the implementation or use of the technology described in

this document or the extent to which any license under such rights

might or might not be available; neither does it represent that it

has made any effort to identify any such rights. Information on the

IETF's procedures with respect to rights in standards-track and

standards-related documentation can be found in BCP-11. Copies of

claims of rights made available for publication and any assurances of

licenses to be made available, or the result of an attempt made to

obtain a general license or permission for the use of such

proprietary rights by implementors or users of this specification can

be obtained from the IETF Secretariat.

The IETF invites any interested party to bring to its attention any

copyrights, patents or patent applications, or other proprietary

rights which may cover technology that may be required to practice

this standard. Please address the information to the IETF Executive

Director.

17. Acknowledgments

This memorandum is based on discussions within the IETF Audio/Video

Transport working group chaired by Stephen Casner and Colin Perkins.

The current protocol has its origins in the Network Voice Protocol

and the Packet Video Protocol (Danny Cohen and Randy Cole) and the

protocol implemented by the vat application (Van Jacobson and Steve

McCanne). Christian Huitema provided ideas for the random identifier

generator. Extensive analysis and simulation of the timer

reconsideration algorithm was done by Jonathan Rosenberg. The

additions for layered encodings were specified by Michael Speer and

Steve McCanne.

Schulzrinne, et al. Standards Track [Page 74]

RFC 3550 RTP July 2003

Appendix A - Algorithms

We provide examples of C code for aspects of RTP sender and receiver

algorithms. There may be other implementation methods that are

faster in particular operating environments or have other advantages.

These implementation notes are for informational purposes only and

are meant to clarify the RTP specification.

The following definitions are used for all examples; for clarity and

brevity, the structure definitions are only valid for 32-bit bigendian

(most significant octet first) architectures. Bit fields are

assumed to be packed tightly in big-endian bit order, with no

additional padding. Modifications would be required to construct a

portable implementation.

/*

* rtp.h -- RTP header file

*/

#include <sys/types.h>

/*

* The type definitions below are valid for 32-bit architectures and

* may have to be adjusted for 16- or 64-bit architectures.

*/

typedef unsigned char u_int8;

typedef unsigned short u_int16;

typedef unsigned int u_int32;

typedef short int16;

/*

* Current protocol version.

*/

#define RTP_VERSION 2

#define RTP_SEQ_MOD (1<<16)

#define RTP_MAX_SDES 255 /* maximum text length for SDES */

typedef enum {

RTCP_SR = 200,

RTCP_RR = 201,

RTCP_SDES = 202,

RTCP_BYE = 203,

RTCP_APP = 204

} rtcp_type_t;

typedef enum {

RTCP_SDES_END = 0,

RTCP_SDES_CNAME = 1,

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RTCP_SDES_NAME = 2,

RTCP_SDES_EMAIL = 3,

RTCP_SDES_PHONE = 4,

RTCP_SDES_LOC = 5,

RTCP_SDES_TOOL = 6,

RTCP_SDES_NOTE = 7,

RTCP_SDES_PRIV = 8

} rtcp_sdes_type_t;

/*

* RTP data header

*/

typedef struct {

unsigned int version:2; /* protocol version */

unsigned int p:1; /* padding flag */

unsigned int x:1; /* header extension flag */

unsigned int cc:4; /* CSRC count */

unsigned int m:1; /* marker bit */

unsigned int pt:7; /* payload type */

unsigned int seq:16; /* sequence number */

u_int32 ts; /* timestamp */

u_int32 ssrc; /* synchronization source */

u_int32 csrc[1]; /* optional CSRC list */

} rtp_hdr_t;

/*

* RTCP common header word

*/

typedef struct {

unsigned int version:2; /* protocol version */

unsigned int p:1; /* padding flag */

unsigned int count:5; /* varies by packet type */

unsigned int pt:8; /* RTCP packet type */

u_int16 length; /* pkt len in words, w/o this word */

} rtcp_common_t;

/*

* Big-endian mask for version, padding bit and packet type pair

*/

#define RTCP_VALID_MASK (0xc000 | 0x2000 | 0xfe)

#define RTCP_VALID_VALUE ((RTP_VERSION << 14) | RTCP_SR)

/*

* Reception report block

*/

typedef struct {

u_int32 ssrc; /* data source being reported */

unsigned int fraction:8; /* fraction lost since last SR/RR */

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RFC 3550 RTP July 2003

int lost:24; /* cumul. no. pkts lost (signed!) */

u_int32 last_seq; /* extended last seq. no. received */

u_int32 jitter; /* interarrival jitter */

u_int32 lsr; /* last SR packet from this source */

u_int32 dlsr; /* delay since last SR packet */

} rtcp_rr_t;

/*

* SDES item

*/

typedef struct {

u_int8 type; /* type of item (rtcp_sdes_type_t) */

u_int8 length; /* length of item (in octets) */

char data[1]; /* text, not null-terminated */

} rtcp_sdes_item_t;

/*

* One RTCP packet

*/

typedef struct {

rtcp_common_t common; /* common header */

union {

/* sender report (SR) */

struct {

u_int32 ssrc; /* sender generating this report */

u_int32 ntp_sec; /* NTP timestamp */

u_int32 ntp_frac;

u_int32 rtp_ts; /* RTP timestamp */

u_int32 psent; /* packets sent */

u_int32 osent; /* octets sent */

rtcp_rr_t rr[1]; /* variable-length list */

} sr;

/* reception report (RR) */

struct {

u_int32 ssrc; /* receiver generating this report */

rtcp_rr_t rr[1]; /* variable-length list */

} rr;

/* source description (SDES) */

struct rtcp_sdes {

u_int32 src; /* first SSRC/CSRC */

rtcp_sdes_item_t item[1]; /* list of SDES items */

} sdes;

/* BYE */

struct {

u_int32 src[1]; /* list of sources */

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RFC 3550 RTP July 2003

/* can't express trailing text for reason */

} bye;

} r;

} rtcp_t;

typedef struct rtcp_sdes rtcp_sdes_t;

/*

* Per-source state information

*/

typedef struct {

u_int16 max_seq; /* highest seq. number seen */

u_int32 cycles; /* shifted count of seq. number cycles */

u_int32 base_seq; /* base seq number */

u_int32 bad_seq; /* last 'bad' seq number + 1 */

u_int32 probation; /* sequ. packets till source is valid */

u_int32 received; /* packets received */

u_int32 expected_prior; /* packet expected at last interval */

u_int32 received_prior; /* packet received at last interval */

u_int32 transit; /* relative trans time for prev pkt */

u_int32 jitter; /* estimated jitter */

/* ... */

} source;

A.1 RTP Data Header Validity Checks

An RTP receiver should check the validity of the RTP header on

incoming packets since they might be encrypted or might be from a

different application that happens to be misaddressed. Similarly, if

encryption according to the method described in Section 9 is enabled,

the header validity check is needed to verify that incoming packets

have been correctly decrypted, although a failure of the header

validity check (e.g., unknown payload type) may not necessarily

indicate decryption failure.

Only weak validity checks are possible on an RTP data packet from a

source that has not been heard before:

o RTP version field must equal 2.

o The payload type must be known, and in particular it must not be

equal to SR or RR.

o If the P bit is set, then the last octet of the packet must

contain a valid octet count, in particular, less than the total

packet length minus the header size.

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RFC 3550 RTP July 2003

o The X bit must be zero if the profile does not specify that the

header extension mechanism may be used. Otherwise, the extension

length field must be less than the total packet size minus the

fixed header length and padding.

o The length of the packet must be consistent with CC and payload

type (if payloads have a known length).

The last three checks are somewhat complex and not always possible,

leaving only the first two which total just a few bits. If the SSRC

identifier in the packet is one that has been received before, then

the packet is probably valid and checking if the sequence number is

in the expected range provides further validation. If the SSRC

identifier has not been seen before, then data packets carrying that

identifier may be considered invalid until a small number of them

arrive with consecutive sequence numbers. Those invalid packets MAY

be discarded or they MAY be stored and delivered once validation has

been achieved if the resulting delay is acceptable.

The routine update_seq shown below ensures that a source is declared

valid only after MIN_SEQUENTIAL packets have been received in

sequence. It also validates the sequence number seq of a newly

received packet and updates the sequence state for the packet's

source in the structure to which s points.

When a new source is heard for the first time, that is, its SSRC

identifier is not in the table (see Section 8.2), and the per-source

state is allocated for it, s->probation is set to the number of

sequential packets required before declaring a source valid

(parameter MIN_SEQUENTIAL) and other variables are initialized:

init_seq(s, seq);

s->max_seq = seq - 1;

s->probation = MIN_SEQUENTIAL;

A non-zero s->probation marks the source as not yet valid so the

state may be discarded after a short timeout rather than a long one,

as discussed in Section 6.2.1.

After a source is considered valid, the sequence number is considered

valid if it is no more than MAX_DROPOUT ahead of s->max_seq nor more

than MAX_MISORDER behind. If the new sequence number is ahead of

max_seq modulo the RTP sequence number range (16 bits), but is

smaller than max_seq, it has wrapped around and the (shifted) count

of sequence number cycles is incremented. A value of one is returned

to indicate a valid sequence number.

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Otherwise, the value zero is returned to indicate that the validation

failed, and the bad sequence number plus 1 is stored. If the next

packet received carries the next higher sequence number, it is

considered the valid start of a new packet sequence presumably caused

by an extended dropout or a source restart. Since multiple complete

sequence number cycles may have been missed, the packet loss

statistics are reset.

Typical values for the parameters are shown, based on a maximum

misordering time of 2 seconds at 50 packets/second and a maximum

dropout of 1 minute. The dropout parameter MAX_DROPOUT should be a

small fraction of the 16-bit sequence number space to give a

reasonable probability that new sequence numbers after a restart will

not fall in the acceptable range for sequence numbers from before the

restart.

void init_seq(source *s, u_int16 seq)

{

s->base_seq = seq;

s->max_seq = seq;

s->bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */

s->cycles = 0;

s->received = 0;

s->received_prior = 0;

s->expected_prior = 0;

/* other initialization */

}

int update_seq(source *s, u_int16 seq)

{

u_int16 udelta = seq - s->max_seq;

const int MAX_DROPOUT = 3000;

const int MAX_MISORDER = 100;

const int MIN_SEQUENTIAL = 2;

/*

* Source is not valid until MIN_SEQUENTIAL packets with

* sequential sequence numbers have been received.

*/

if (s->probation) {

/* packet is in sequence */

if (seq == s->max_seq + 1) {

s->probation--;

s->max_seq = seq;

if (s->probation == 0) {

init_seq(s, seq);

s->received++;

return 1;

Schulzrinne, et al. Standards Track [Page 80]

RFC 3550 RTP July 2003

}

} else {

s->probation = MIN_SEQUENTIAL - 1;

s->max_seq = seq;

}

return 0;

} else if (udelta < MAX_DROPOUT) {

/* in order, with permissible gap */

if (seq < s->max_seq) {

/*

* Sequence number wrapped - count another 64K cycle.

*/

s->cycles += RTP_SEQ_MOD;

}

s->max_seq = seq;

} else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {

/* the sequence number made a very large jump */

if (seq == s->bad_seq) {

/*

* Two sequential packets -- assume that the other side

* restarted without telling us so just re-sync

* (i.e., pretend this was the first packet).

*/

init_seq(s, seq);

}

else {

s->bad_seq = (seq + 1) & (RTP_SEQ_MOD-1);

return 0;

}

} else {

/* duplicate or reordered packet */

}

s->received++;

return 1;

}

The validity check can be made stronger requiring more than two

packets in sequence. The disadvantages are that a larger number of

initial packets will be discarded (or delayed in a queue) and that

high packet loss rates could prevent validation. However, because

the RTCP header validation is relatively strong, if an RTCP packet is

received from a source before the data packets, the count could be

adjusted so that only two packets are required in sequence. If

initial data loss for a few seconds can be tolerated, an application

MAY choose to discard all data packets from a source until a valid

RTCP packet has been received from that source.

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Depending on the application and encoding, algorithms may exploit

additional knowledge about the payload format for further validation.

For payload types where the timestamp increment is the same for all

packets, the timestamp values can be predicted from the previous

packet received from the same source using the sequence number

difference (assuming no change in payload type).

A strong "fast-path" check is possible since with high probability

the first four octets in the header of a newly received RTP data

packet will be just the same as that of the previous packet from the

same SSRC except that the sequence number will have increased by one.

Similarly, a single-entry cache may be used for faster SSRC lookups

in applications where data is typically received from one source at a

time.

A.2 RTCP Header Validity Checks

The following checks should be applied to RTCP packets.

o RTP version field must equal 2.

o The payload type field of the first RTCP packet in a compound

packet must be equal to SR or RR.

o The padding bit (P) should be zero for the first packet of a

compound RTCP packet because padding should only be applied, if it

is needed, to the last packet.

o The length fields of the individual RTCP packets must add up to

the overall length of the compound RTCP packet as received. This

is a fairly strong check.

The code fragment below performs all of these checks. The packet

type is not checked for subsequent packets since unknown packet types

may be present and should be ignored.

u_int32 len; /* length of compound RTCP packet in words */

rtcp_t *r; /* RTCP header */

rtcp_t *end; /* end of compound RTCP packet */

if ((*(u_int16 *)r & RTCP_VALID_MASK) != RTCP_VALID_VALUE) {

/* something wrong with packet format */

}

end = (rtcp_t *)((u_int32 *)r + len);

do r = (rtcp_t *)((u_int32 *)r + r->common.length + 1);

while (r < end && r->common.version == 2);

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RFC 3550 RTP July 2003

if (r != end) {

/* something wrong with packet format */

}

A.3 Determining Number of Packets Expected and Lost

In order to compute packet loss rates, the number of RTP packets

expected and actually received from each source needs to be known,

using per-source state information defined in struct source

referenced via pointer s in the code below. The number of packets

received is simply the count of packets as they arrive, including any

late or duplicate packets. The number of packets expected can be

computed by the receiver as the difference between the highest

sequence number received (s->max_seq) and the first sequence number

received (s->base_seq). Since the sequence number is only 16 bits

and will wrap around, it is necessary to extend the highest sequence

number with the (shifted) count of sequence number wraparounds

(s->cycles). Both the received packet count and the count of cycles

are maintained the RTP header validity check routine in Appendix A.1.

extended_max = s->cycles + s->max_seq;

expected = extended_max - s->base_seq + 1;

The number of packets lost is defined to be the number of packets

expected less the number of packets actually received:

lost = expected - s->received;

Since this signed number is carried in 24 bits, it should be clamped

at 0x7fffff for positive loss or 0x800000 for negative loss rather

than wrapping around.

The fraction of packets lost during the last reporting interval

(since the previous SR or RR packet was sent) is calculated from

differences in the expected and received packet counts across the

interval, where expected_prior and received_prior are the values

saved when the previous reception report was generated:

expected_interval = expected - s->expected_prior;

s->expected_prior = expected;

received_interval = s->received - s->received_prior;

s->received_prior = s->received;

lost_interval = expected_interval - received_interval;

if (expected_interval == 0 || lost_interval <= 0) fraction = 0;

else fraction = (lost_interval << 8) / expected_interval;

The resulting fraction is an 8-bit fixed point number with the binary

point at the left edge.

Schulzrinne, et al. Standards Track [Page 83]

RFC 3550 RTP July 2003

A.4 Generating RTCP SDES Packets

This function builds one SDES chunk into buffer b composed of argc

items supplied in arrays type, value and length. It returns a

pointer to the next available location within b.

char *rtp_write_sdes(char *b, u_int32 src, int argc,

rtcp_sdes_type_t type[], char *value[],

int length[])

{

rtcp_sdes_t *s = (rtcp_sdes_t *)b;

rtcp_sdes_item_t *rsp;

int i;

int len;

int pad;

/* SSRC header */

s->src = src;

rsp = &s->item[0];

/* SDES items */

for (i = 0; i < argc; i++) {

rsp->type = type[i];

len = length[i];

if (len > RTP_MAX_SDES) {

/* invalid length, may want to take other action */

len = RTP_MAX_SDES;

}

rsp->length = len;

memcpy(rsp->data, value[i], len);

rsp = (rtcp_sdes_item_t *)&rsp->data[len];

}

/* terminate with end marker and pad to next 4-octet boundary */

len = ((char *) rsp) - b;

pad = 4 - (len & 0x3);

b = (char *) rsp;

while (pad--) *b++ = RTCP_SDES_END;

return b;

}

Schulzrinne, et al. Standards Track [Page 84]

RFC 3550 RTP July 2003

A.5 Parsing RTCP SDES Packets

This function parses an SDES packet, calling functions find_member()

to find a pointer to the information for a session member given the

SSRC identifier and member_sdes() to store the new SDES information

for that member. This function expects a pointer to the header of

the RTCP packet.

void rtp_read_sdes(rtcp_t *r)

{

int count = r->common.count;

rtcp_sdes_t *sd = &r->r.sdes;

rtcp_sdes_item_t *rsp, *rspn;

rtcp_sdes_item_t *end = (rtcp_sdes_item_t *)

((u_int32 *)r + r->common.length + 1);

source *s;

while (--count >= 0) {

rsp = &sd->item[0];

if (rsp >= end) break;

s = find_member(sd->src);

for (; rsp->type; rsp = rspn ) {

rspn = (rtcp_sdes_item_t *)((char*)rsp+rsp->length+2);

if (rspn >= end) {

rsp = rspn;

break;

}

member_sdes(s, rsp->type, rsp->data, rsp->length);

}

sd = (rtcp_sdes_t *)

((u_int32 *)sd + (((char *)rsp - (char *)sd) >> 2)+1);

}

if (count >= 0) {

/* invalid packet format */

}

}

A.6 Generating a Random 32-bit Identifier

The following subroutine generates a random 32-bit identifier using

the MD5 routines published in RFC 1321 [32]. The system routines may

not be present on all operating systems, but they should serve as

hints as to what kinds of information may be used. Other system

calls that may be appropriate include

Schulzrinne, et al. Standards Track [Page 85]

RFC 3550 RTP July 2003

o getdomainname(),

o getwd(), or

o getrusage().

"Live" video or audio samples are also a good source of random

numbers, but care must be taken to avoid using a turned-off

microphone or blinded camera as a source [17].

Use of this or a similar routine is recommended to generate the

initial seed for the random number generator producing the RTCP

period (as shown in Appendix A.7), to generate the initial values for

the sequence number and timestamp, and to generate SSRC values.

Since this routine is likely to be CPU-intensive, its direct use to

generate RTCP periods is inappropriate because predictability is not

an issue. Note that this routine produces the same result on

repeated calls until the value of the system clock changes unless

different values are supplied for the type argument.

/*

* Generate a random 32-bit quantity.

*/

#include <sys/types.h> /* u_long */

#include <sys/time.h> /* gettimeofday() */

#include <unistd.h> /* get..() */

#include <stdio.h> /* printf() */

#include <time.h> /* clock() */

#include <sys/utsname.h> /* uname() */

#include "global.h" /* from RFC 1321 */

#include "md5.h" /* from RFC 1321 */

#define MD_CTX MD5_CTX

#define MDInit MD5Init

#define MDUpdate MD5Update

#define MDFinal MD5Final

static u_long md_32(char *string, int length)

{

MD_CTX context;

union {

char c[16];

u_long x[4];

} digest;

u_long r;

int i;

MDInit (&context);

Schulzrinne, et al. Standards Track [Page 86]

RFC 3550 RTP July 2003

MDUpdate (&context, string, length);

MDFinal ((unsigned char *)&digest, &context);

r = 0;

for (i = 0; i < 3; i++) {

r ^= digest.x[i];

}

return r;

} /* md_32 */

/*

* Return random unsigned 32-bit quantity. Use 'type' argument if

* you need to generate several different values in close succession.

*/

u_int32 random32(int type)

{

struct {

int type;

struct timeval tv;

clock_t cpu;

pid_t pid;

u_long hid;

uid_t uid;

gid_t gid;

struct utsname name;

} s;

gettimeofday(&s.tv, 0);

uname(&s.name);

s.type = type;

s.cpu = clock();

s.pid = getpid();

s.hid = gethostid();

s.uid = getuid();

s.gid = getgid();

/* also: system uptime */

return md_32((char *)&s, sizeof(s));

} /* random32 */

A.7 Computing the RTCP Transmission Interval

The following functions implement the RTCP transmission and reception

rules described in Section 6.2. These rules are coded in several

functions:

o rtcp_interval() computes the deterministic calculated interval,

measured in seconds. The parameters are defined in Section 6.3.

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RFC 3550 RTP July 2003

o OnExpire() is called when the RTCP transmission timer expires.

o OnReceive() is called whenever an RTCP packet is received.

Both OnExpire() and OnReceive() have event e as an argument. This is

the next scheduled event for that participant, either an RTCP report

or a BYE packet. It is assumed that the following functions are

available:

o Schedule(time t, event e) schedules an event e to occur at time t.

When time t arrives, the function OnExpire is called with e as an

argument.

o Reschedule(time t, event e) reschedules a previously scheduled

event e for time t.

o SendRTCPReport(event e) sends an RTCP report.

o SendBYEPacket(event e) sends a BYE packet.

o TypeOfEvent(event e) returns EVENT_BYE if the event being

processed is for a BYE packet to be sent, else it returns

EVENT_REPORT.

o PacketType(p) returns PACKET_RTCP_REPORT if packet p is an RTCP

report (not BYE), PACKET_BYE if its a BYE RTCP packet, and

PACKET_RTP if its a regular RTP data packet.

o ReceivedPacketSize() and SentPacketSize() return the size of the

referenced packet in octets.

o NewMember(p) returns a 1 if the participant who sent packet p is

not currently in the member list, 0 otherwise. Note this function

is not sufficient for a complete implementation because each CSRC

identifier in an RTP packet and each SSRC in a BYE packet should

be processed.

o NewSender(p) returns a 1 if the participant who sent packet p is

not currently in the sender sublist of the member list, 0

otherwise.

o AddMember() and RemoveMember() to add and remove participants from

the member list.

o AddSender() and RemoveSender() to add and remove participants from

the sender sublist of the member list.

Schulzrinne, et al. Standards Track [Page 88]

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These functions would have to be extended for an implementation that

allows the RTCP bandwidth fractions for senders and non-senders to be

specified as explicit parameters rather than fixed values of 25% and

75%. The extended implementation of rtcp_interval() would need to

avoid division by zero if one of the parameters was zero.

double rtcp_interval(int members,

int senders,

double rtcp_bw,

int we_sent,

double avg_rtcp_size,

int initial)

{

/*

* Minimum average time between RTCP packets from this site (in

* seconds). This time prevents the reports from `clumping' when

* sessions are small and the law of large numbers isn't helping

* to smooth out the traffic. It also keeps the report interval

* from becoming ridiculously small during transient outages like

* a network partition.

*/

double const RTCP_MIN_TIME = 5.;

/*

* Fraction of the RTCP bandwidth to be shared among active

* senders. (This fraction was chosen so that in a typical

* session with one or two active senders, the computed report

* time would be roughly equal to the minimum report time so that

* we don't unnecessarily slow down receiver reports.) The

* receiver fraction must be 1 - the sender fraction.

*/

double const RTCP_SENDER_BW_FRACTION = 0.25;

double const RTCP_RCVR_BW_FRACTION = (1-RTCP_SENDER_BW_FRACTION);

/*

/* To compensate for "timer reconsideration" converging to a

* value below the intended average.

*/

double const COMPENSATION = 2.71828 - 1.5;

double t; /* interval */

double rtcp_min_time = RTCP_MIN_TIME;

int n; /* no. of members for computation */

/*

* Very first call at application start-up uses half the min

* delay for quicker notification while still allowing some time

* before reporting for randomization and to learn about other

* sources so the report interval will converge to the correct

* interval more quickly.

Schulzrinne, et al. Standards Track [Page 89]

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*/

if (initial) {

rtcp_min_time /= 2;

}

/*

* Dedicate a fraction of the RTCP bandwidth to senders unless

* the number of senders is large enough that their share is

* more than that fraction.

*/

n = members;

if (senders <= members * RTCP_SENDER_BW_FRACTION) {

if (we_sent) {

rtcp_bw *= RTCP_SENDER_BW_FRACTION;

n = senders;

} else {

rtcp_bw *= RTCP_RCVR_BW_FRACTION;

n -= senders;

}

}

/*

* The effective number of sites times the average packet size is

* the total number of octets sent when each site sends a report.

* Dividing this by the effective bandwidth gives the time

* interval over which those packets must be sent in order to

* meet the bandwidth target, with a minimum enforced. In that

* time interval we send one report so this time is also our

* average time between reports.

*/

t = avg_rtcp_size * n / rtcp_bw;

if (t < rtcp_min_time) t = rtcp_min_time;

/*

* To avoid traffic bursts from unintended synchronization with

* other sites, we then pick our actual next report interval as a

* random number uniformly distributed between 0.5*t and 1.5*t.

*/

t = t * (drand48() + 0.5);

t = t / COMPENSATION;

return t;

}

void OnExpire(event e,

int members,

int senders,

double rtcp_bw,

int we_sent,

double *avg_rtcp_size,

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int *initial,

time_tp tc,

time_tp *tp,

int *pmembers)

{

/* This function is responsible for deciding whether to send an

* RTCP report or BYE packet now, or to reschedule transmission.

* It is also responsible for updating the pmembers, initial, tp,

* and avg_rtcp_size state variables. This function should be

* called upon expiration of the event timer used by Schedule().

*/

double t; /* Interval */

double tn; /* Next transmit time */

/* In the case of a BYE, we use "timer reconsideration" to

* reschedule the transmission of the BYE if necessary */

if (TypeOfEvent(e) == EVENT_BYE) {

t = rtcp_interval(members,

senders,

rtcp_bw,

we_sent,

*avg_rtcp_size,

*initial);

tn = *tp + t;

if (tn <= tc) {

SendBYEPacket(e);

exit(1);

} else {

Schedule(tn, e);

}

} else if (TypeOfEvent(e) == EVENT_REPORT) {

t = rtcp_interval(members,

senders,

rtcp_bw,

we_sent,

*avg_rtcp_size,

*initial);

tn = *tp + t;

if (tn <= tc) {

SendRTCPReport(e);

*avg_rtcp_size = (1./16.)*SentPacketSize(e) +

(15./16.)*(*avg_rtcp_size);

*tp = tc;

/* We must redraw the interval. Don't reuse the

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RFC 3550 RTP July 2003

one computed above, since its not actually

distributed the same, as we are conditioned

on it being small enough to cause a packet to

be sent */

t = rtcp_interval(members,

senders,

rtcp_bw,

we_sent,

*avg_rtcp_size,

*initial);

Schedule(t+tc,e);

*initial = 0;

} else {

Schedule(tn, e);

}

*pmembers = members;

}

}

void OnReceive(packet p,

event e,

int *members,

int *pmembers,

int *senders,

double *avg_rtcp_size,

double *tp,

double tc,

double tn)

{

/* What we do depends on whether we have left the group, and are

* waiting to send a BYE (TypeOfEvent(e) == EVENT_BYE) or an RTCP

* report. p represents the packet that was just received. */

if (PacketType(p) == PACKET_RTCP_REPORT) {

if (NewMember(p) && (TypeOfEvent(e) == EVENT_REPORT)) {

AddMember(p);

*members += 1;

}

*avg_rtcp_size = (1./16.)*ReceivedPacketSize(p) +

(15./16.)*(*avg_rtcp_size);

} else if (PacketType(p) == PACKET_RTP) {

if (NewMember(p) && (TypeOfEvent(e) == EVENT_REPORT)) {

AddMember(p);

*members += 1;

}

if (NewSender(p) && (TypeOfEvent(e) == EVENT_REPORT)) {

Schulzrinne, et al. Standards Track [Page 92]

RFC 3550 RTP July 2003

AddSender(p);

*senders += 1;

}

} else if (PacketType(p) == PACKET_BYE) {

*avg_rtcp_size = (1./16.)*ReceivedPacketSize(p) +

(15./16.)*(*avg_rtcp_size);

if (TypeOfEvent(e) == EVENT_REPORT) {

if (NewSender(p) == FALSE) {

RemoveSender(p);

*senders -= 1;

}

if (NewMember(p) == FALSE) {

RemoveMember(p);

*members -= 1;

}

if (*members < *pmembers) {

tn = tc +

(((double) *members)/(*pmembers))*(tn - tc);

*tp = tc -

(((double) *members)/(*pmembers))*(tc - *tp);

/* Reschedule the next report for time tn */

Reschedule(tn, e);

*pmembers = *members;

}

} else if (TypeOfEvent(e) == EVENT_BYE) {

*members += 1;

}

}

}

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A.8 Estimating the Interarrival Jitter

The code fragments below implement the algorithm given in Section

6.4.1 for calculating an estimate of the statistical variance of the

RTP data interarrival time to be inserted in the interarrival jitter

field of reception reports. The inputs are r->ts, the timestamp from

the incoming packet, and arrival, the current time in the same units.

Here s points to state for the source; s->transit holds the relative

transit time for the previous packet, and s->jitter holds the

estimated jitter. The jitter field of the reception report is

measured in timestamp units and expressed as an unsigned integer, but

the jitter estimate is kept in a floating point. As each data packet

arrives, the jitter estimate is updated:

int transit = arrival - r->ts;

int d = transit - s->transit;

s->transit = transit;

if (d < 0) d = -d;

s->jitter += (1./16.) * ((double)d - s->jitter);

When a reception report block (to which rr points) is generated for

this member, the current jitter estimate is returned:

rr->jitter = (u_int32) s->jitter;

Alternatively, the jitter estimate can be kept as an integer, but

scaled to reduce round-off error. The calculation is the same except

for the last line:

s->jitter += d - ((s->jitter + 8) >> 4);

In this case, the estimate is sampled for the reception report as:

rr->jitter = s->jitter >> 4;

Schulzrinne, et al. Standards Track [Page 94]

RFC 3550 RTP July 2003

Appendix B - Changes from RFC 1889

Most of this RFC is identical to RFC 1889. There are no changes in

the packet formats on the wire, only changes to the rules and

algorithms governing how the protocol is used. The biggest change is

an enhancement to the scalable timer algorithm for calculating when

to send RTCP packets:

o The algorithm for calculating the RTCP transmission interval

specified in Sections 6.2 and 6.3 and illustrated in Appendix A.7

is augmented to include "reconsideration" to minimize transmission

in excess of the intended rate when many participants join a

session simultaneously, and "reverse reconsideration" to reduce

the incidence and duration of false participant timeouts when the

number of participants drops rapidly. Reverse reconsideration is

also used to possibly shorten the delay before sending RTCP SR

when transitioning from passive receiver to active sender mode.

o Section 6.3.7 specifies new rules controlling when an RTCP BYE

packet should be sent in order to avoid a flood of packets when

many participants leave a session simultaneously.

o The requirement to retain state for inactive participants for a

period long enough to span typical network partitions was removed

from Section 6.2.1. In a session where many participants join for

a brief time and fail to send BYE, this requirement would cause a

significant overestimate of the number of participants. The

reconsideration algorithm added in this revision compensates for

the large number of new participants joining simultaneously when a

partition heals.

It should be noted that these enhancements only have a significant

effect when the number of session participants is large (thousands)

and most of the participants join or leave at the same time. This

makes testing in a live network difficult. However, the algorithm

was subjected to a thorough analysis and simulation to verify its

performance. Furthermore, the enhanced algorithm was designed to

interoperate with the algorithm in RFC 1889 such that the degree of

reduction in excess RTCP bandwidth during a step join is proportional

to the fraction of participants that implement the enhanced

algorithm. Interoperation of the two algorithms has been verified

experimentally on live networks.

Other functional changes were:

o Section 6.2.1 specifies that implementations may store only a

sampling of the participants' SSRC identifiers to allow scaling to

very large sessions. Algorithms are specified in RFC 2762 [21].

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RFC 3550 RTP July 2003

o In Section 6.2 it is specified that RTCP sender and non-sender

bandwidths may be set as separate parameters of the session rather

than a strict percentage of the session bandwidth, and may be set

to zero. The requirement that RTCP was mandatory for RTP sessions

using IP multicast was relaxed. However, a clarification was also

added that turning off RTCP is NOT RECOMMENDED.

o In Sections 6.2, 6.3.1 and Appendix A.7, it is specified that the

fraction of participants below which senders get dedicated RTCP

bandwidth changes from the fixed 1/4 to a ratio based on the RTCP

sender and non-sender bandwidth parameters when those are given.

The condition that no bandwidth is dedicated to senders when there

are no senders was removed since that is expected to be a

transitory state. It also keeps non-senders from using sender

RTCP bandwidth when that is not intended.

o Also in Section 6.2 it is specified that the minimum RTCP interval

may be scaled to smaller values for high bandwidth sessions, and

that the initial RTCP delay may be set to zero for unicast

sessions.

o Timing out a participant is to be based on inactivity for a number

of RTCP report intervals calculated using the receiver RTCP

bandwidth fraction even for active senders.

o Sections 7.2 and 7.3 specify that translators and mixers should

send BYE packets for the sources they are no longer forwarding.

o Rule changes for layered encodings are defined in Sections 2.4,

6.3.9, 8.3 and 11. In the last of these, it is noted that the

address and port assignment rule conflicts with the SDP

specification, RFC 2327 [15], but it is intended that this

restriction will be relaxed in a revision of RFC 2327.

o The convention for using even/odd port pairs for RTP and RTCP in

Section 11 was clarified to refer to destination ports. The

requirement to use an even/odd port pair was removed if the two

ports are specified explicitly. For unicast RTP sessions,

distinct port pairs may be used for the two ends (Sections 3, 7.1

and 11).

o A new Section 10 was added to explain the requirement for

congestion control in applications using RTP.

o In Section 8.2, the requirement that a new SSRC identifier MUST be

chosen whenever the source transport address is changed has been

relaxed to say that a new SSRC identifier MAY be chosen.

Correspondingly, it was clarified that an implementation MAY

Schulzrinne, et al. Standards Track [Page 96]

RFC 3550 RTP July 2003

choose to keep packets from the new source address rather than the

existing source address when an SSRC collision occurs between two

other participants, and SHOULD do so for applications such as

telephony in which some sources such as mobile entities may change

addresses during the course of an RTP session.

o An indentation bug in the RFC 1889 printing of the pseudo-code for

the collision detection and resolution algorithm in Section 8.2

has been corrected by translating the syntax to pseudo C language,

and the algorithm has been modified to remove the restriction that

both RTP and RTCP must be sent from the same source port number.

o The description of the padding mechanism for RTCP packets was

clarified and it is specified that padding MUST only be applied to

the last packet of a compound RTCP packet.

o In Section A.1, initialization of base_seq was corrected to be seq

rather than seq - 1, and the text was corrected to say the bad

sequence number plus 1 is stored. The initialization of max_seq

and other variables for the algorithm was separated from the text

to make clear that this initialization must be done in addition to

calling the init_seq() function (and a few words lost in RFC 1889

when processing the document from source to output form were

restored).

o Clamping of number of packets lost in Section A.3 was corrected to

use both positive and negative limits.

o The specification of "relative" NTP timestamp in the RTCP SR

section now defines these timestamps to be based on the most

common system-specific clock, such as system uptime, rather than

on session elapsed time which would not be the same for multiple

applications started on the same machine at different times.

Non-functional changes:

o It is specified that a receiver MUST ignore packets with payload

types it does not understand.

o In Fig. 2, the floating point NTP timestamp value was corrected,

some missing leading zeros were added in a hex number, and the UTC

timezone was specified.

o The inconsequence of NTP timestamps wrapping around in the year

2036 is explained.

Schulzrinne, et al. Standards Track [Page 97]

RFC 3550 RTP July 2003

o The policy for registration of RTCP packet types and SDES types

was clarified in a new Section 15, IANA Considerations. The

suggestion that experimenters register the numbers they need and

then unregister those which prove to be unneeded has been removed

in favor of using APP and PRIV. Registration of profile names was

also specified.

o The reference for the UTF-8 character set was changed from an

X/Open Preliminary Specification to be RFC 2279.

o The reference for RFC 1597 was updated to RFC 1918 and the

reference for RFC 2543 was updated to RFC 3261.

o The last paragraph of the introduction in RFC 1889, which

cautioned implementors to limit deployment in the Internet, was

removed because it was deemed no longer relevant.

o A non-normative note regarding the use of RTP with Source-Specific

Multicast (SSM) was added in Section 6.

o The definition of "RTP session" in Section 3 was expanded to

acknowledge that a single session may use multiple destination

transport addresses (as was always the case for a translator or

mixer) and to explain that the distinguishing feature of an RTP

session is that each corresponds to a separate SSRC identifier

space. A new definition of "multimedia session" was added to

reduce confusion about the word "session".

o The meaning of "sampling instant" was explained in more detail as

part of the definition of the timestamp field of the RTP header in

Section 5.1.

o Small clarifications of the text have been made in several places,

some in response to questions from readers. In particular:

- In RFC 1889, the first five words of the second sentence of

Section 2.2 were lost in processing the document from source to

output form, but are now restored.

- A definition for "RTP media type" was added in Section 3 to

allow the explanation of multiplexing RTP sessions in Section

5.2 to be more clear regarding the multiplexing of multiple

media. That section also now explains that multiplexing

multiple sources of the same medium based on SSRC identifiers

may be appropriate and is the norm for multicast sessions.

- The definition for "non-RTP means" was expanded to include

examples of other protocols constituting non-RTP means.

Schulzrinne, et al. Standards Track [Page 98]

RFC 3550 RTP July 2003

- The description of the session bandwidth parameter is expanded

in Section 6.2, including a clarification that the control

traffic bandwidth is in addition to the session bandwidth for

the data traffic.

- The effect of varying packet duration on the jitter calculation

was explained in Section 6.4.4.

- The method for terminating and padding a sequence of SDES items

was clarified in Section 6.5.

- IPv6 address examples were added in the description of SDES

CNAME in Section 6.5.1, and "example.com" was used in place of

other example domain names.

- The Security section added a formal reference to IPSEC now that

it is available, and says that the confidentiality method

defined in this specification is primarily to codify existing

practice. It is RECOMMENDED that stronger encryption

algorithms such as Triple-DES be used in place of the default

algorithm, and noted that the SRTP profile based on AES will be

the correct choice in the future. A caution about the weakness

of the RTP header as an initialization vector was added. It

was also noted that payload-only encryption is necessary to

allow for header compression.

- The method for partial encryption of RTCP was clarified; in

particular, SDES CNAME is carried in only one part when the

compound RTCP packet is split.

- It is clarified that only one compound RTCP packet should be

sent per reporting interval and that if there are too many

active sources for the reports to fit in the MTU, then a subset

of the sources should be selected round-robin over multiple

intervals.

- A note was added in Appendix A.1 that packets may be saved

during RTP header validation and delivered upon success.

- Section 7.3 now explains that a mixer aggregating SDES packets

uses more RTCP bandwidth due to longer packets, and a mixer

passing through RTCP naturally sends packets at higher than the

single source rate, but both behaviors are valid.

- Section 13 clarifies that an RTP application may use multiple

profiles but typically only one in a given session.

Schulzrinne, et al. Standards Track [Page 99]

RFC 3550 RTP July 2003

- The terms MUST, SHOULD, MAY, etc. are used as defined in RFC

2119.

- The bibliography was divided into normative and informative

references.

References

Normative References

[1] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video

Conferences with Minimal Control", RFC 3551, July 2003.

[2] Bradner, S., "Key Words for Use in RFCs to Indicate Requirement

Levels", BCP 14, RFC 2119, March 1997.

[3] Postel, J., "Internet Protocol", STD 5, RFC 791, September 1981.

[4] Mills, D., "Network Time Protocol (Version 3) Specification,

Implementation and Analysis", RFC 1305, March 1992.

[5] Yergeau, F., "UTF-8, a Transformation Format of ISO 10646", RFC

2279, January 1998.

[6] Mockapetris, P., "Domain Names - Concepts and Facilities", STD

13, RFC 1034, November 1987.

[7] Mockapetris, P., "Domain Names - Implementation and

Specification", STD 13, RFC 1035, November 1987.

[8] Braden, R., "Requirements for Internet Hosts - Application and

Support", STD 3, RFC 1123, October 1989.

[9] Resnick, P., "Internet Message Format", RFC 2822, April 2001.

Informative References

[10] Clark, D. and D. Tennenhouse, "Architectural Considerations for

a New Generation of Protocols," in SIGCOMM Symposium on

Communications Architectures and Protocols , (Philadelphia,

Pennsylvania), pp. 200--208, IEEE Computer Communications

Review, Vol. 20(4), September 1990.

[11] Schulzrinne, H., "Issues in designing a transport protocol for

audio and video conferences and other multiparticipant real-time

applications." expired Internet Draft, October 1993.

Schulzrinne, et al. Standards Track [Page 100]

RFC 3550 RTP July 2003

[12] Comer, D., Internetworking with TCP/IP , vol. 1. Englewood

Cliffs, New Jersey: Prentice Hall, 1991.

[13] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,

Peterson, J., Sparks, R., Handley, M. and E. Schooler, "SIP:

Session Initiation Protocol", RFC 3261, June 2002.

[14] International Telecommunication Union, "Visual telephone systems

and equipment for local area networks which provide a nonguaranteed

quality of service", Recommendation H.323,

Telecommunication Standardization Sector of ITU, Geneva,

Switzerland, July 2003.

[15] Handley, M. and V. Jacobson, "SDP: Session Description

Protocol", RFC 2327, April 1998.

[16] Schulzrinne, H., Rao, A. and R. Lanphier, "Real Time Streaming

Protocol (RTSP)", RFC 2326, April 1998.

[17] Eastlake 3rd, D., Crocker, S. and J. Schiller, "Randomness

Recommendations for Security", RFC 1750, December 1994.

[18] Bolot, J.-C., Turletti, T. and I. Wakeman, "Scalable Feedback

Control for Multicast Video Distribution in the Internet", in

SIGCOMM Symposium on Communications Architectures and Protocols,

(London, England), pp. 58--67, ACM, August 1994.

[19] Busse, I., Deffner, B. and H. Schulzrinne, "Dynamic QoS Control

of Multimedia Applications Based on RTP", Computer

Communications , vol. 19, pp. 49--58, January 1996.

[20] Floyd, S. and V. Jacobson, "The Synchronization of Periodic

Routing Messages", in SIGCOMM Symposium on Communications

Architectures and Protocols (D. P. Sidhu, ed.), (San Francisco,

California), pp. 33--44, ACM, September 1993. Also in [34].

[21] Rosenberg, J. and H. Schulzrinne, "Sampling of the Group

Membership in RTP", RFC 2762, February 2000.

[22] Cadzow, J., Foundations of Digital Signal Processing and Data

Analysis New York, New York: Macmillan, 1987.

[23] Hinden, R. and S. Deering, "Internet Protocol Version 6 (IPv6)

Addressing Architecture", RFC 3513, April 2003.

[24] Rekhter, Y., Moskowitz, B., Karrenberg, D., de Groot, G. and E.

Lear, "Address Allocation for Private Internets", RFC 1918,

February 1996.

Schulzrinne, et al. Standards Track [Page 101]

RFC 3550 RTP July 2003

[25] Lear, E., Fair, E., Crocker, D. and T. Kessler, "Network 10

Considered Harmful (Some Practices Shouldn't be Codified)", RFC

1627, July 1994.

[26] Feller, W., An Introduction to Probability Theory and its

Applications, vol. 1. New York, New York: John Wiley and Sons,

third ed., 1968.

[27] Kent, S. and R. Atkinson, "Security Architecture for the

Internet Protocol", RFC 2401, November 1998.

[28] Baugher, M., Blom, R., Carrara, E., McGrew, D., Naslund, M.,

Norrman, K. and D. Oran, "Secure Real-time Transport Protocol",

Work in Progress, April 2003.

[29] Balenson, D., "Privacy Enhancement for Internet Electronic Mail:

Part III", RFC 1423, February 1993.

[30] Voydock, V. and S. Kent, "Security Mechanisms in High-Level

Network Protocols", ACM Computing Surveys, vol. 15, pp. 135-171,

June 1983.

[31] Floyd, S., "Congestion Control Principles", BCP 41, RFC 2914,

September 2000.

[32] Rivest, R., "The MD5 Message-Digest Algorithm", RFC 1321, April

1992.

[33] Stubblebine, S., "Security Services for Multimedia

Conferencing", in 16th National Computer Security Conference,

(Baltimore, Maryland), pp. 391--395, September 1993.

[34] Floyd, S. and V. Jacobson, "The Synchronization of Periodic

Routing Messages", IEEE/ACM Transactions on Networking, vol. 2,

pp. 122--136, April 1994.

Schulzrinne, et al. Standards Track [Page 102]

RFC 3550 RTP July 2003

Authors' Addresses

Henning Schulzrinne

Department of Computer Science

Columbia University

1214 Amsterdam Avenue

New York, NY 10027

United States

EMail: [e-mail]

Stephen L. Casner

Packet Design

3400 Hillview Avenue, Building 3

Palo Alto, CA 94304

United States

EMail: [e-mail]

Ron Frederick

Blue Coat Systems Inc.

650 Almanor Avenue

Sunnyvale, CA 94085

United States

EMail: [e-mail]

Van Jacobson

Packet Design

3400 Hillview Avenue, Building 3

Palo Alto, CA 94304

United States

EMail: [e-mail]

Schulzrinne, et al. Standards Track [Page 103]

RFC 3550 RTP July 2003

Full Copyright Statement

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Schulzrinne, et al. Standards Track [Page 104]