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rtsp协议相关之-rfc1889(RTP 实时应用传送协议文档).txt

王朝other·作者佚名  2006-01-09
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Network Working Group Audio-Video Transport Working Group

Request for Comments: 1889 H. Schulzrinne

Category: Standards Track GMD Fokus

S. Casner

Precept Software, Inc.

R. Frederick

Xerox Palo Alto Research Center

V. Jacobson

Lawrence Berkeley National Laboratory

January 1996

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.

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.

Table of Contents

1. Introduction ........................................ 3

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

2.1 Simple Multicast Audio Conference ................... 5

2.2 Audio and Video Conference .......................... 6

2.3 Mixers and Translators .............................. 6

3. Definitions ......................................... 7

4. Byte Order, Alignment, and Time Format .............. 9

5. RTP Data Transfer Protocol .......................... 10

5.1 RTP Fixed Header Fields ............................. 10

5.2 Multiplexing RTP Sessions ........................... 13

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RFC 1889 RTP January 1996

5.3 Profile-Specific Modifications to the RTP Header..... 14

5.3.1 RTP Header Extension ................................ 14

6. RTP Control Protocol -- RTCP ........................ 15

6.1 RTCP Packet Format .................................. 17

6.2 RTCP Transmission Interval .......................... 19

6.2.1 Maintaining the number of session members ........... 21

6.2.2 Allocation of source description bandwidth .......... 21

6.3 Sender and Receiver Reports ......................... 22

6.3.1 SR: Sender report RTCP packet ....................... 23

6.3.2 RR: Receiver report RTCP packet ..................... 28

6.3.3 Extending the sender and receiver reports ........... 29

6.3.4 Analyzing sender and receiver reports ............... 29

6.4 SDES: Source description RTCP packet ................ 31

6.4.1 CNAME: Canonical end-point identifier SDES item ..... 32

6.4.2 NAME: User name SDES item ........................... 34

6.4.3 EMAIL: Electronic mail address SDES item ............ 34

6.4.4 PHONE: Phone number SDES item ....................... 34

6.4.5 LOC: Geographic user location SDES item ............. 35

6.4.6 TOOL: Application or tool name SDES item ............ 35

6.4.7 NOTE: Notice/status SDES item ....................... 35

6.4.8 PRIV: Private extensions SDES item .................. 36

6.5 BYE: Goodbye RTCP packet ............................ 37

6.6 APP: Application-defined RTCP packet ................ 38

7. RTP Translators and Mixers .......................... 39

7.1 General Description ................................. 39

7.2 RTCP Processing in Translators ...................... 41

7.3 RTCP Processing in Mixers ........................... 43

7.4 Cascaded Mixers ..................................... 44

8. SSRC Identifier Allocation and Use .................. 44

8.1 Probability of Collision ............................ 44

8.2 Collision Resolution and Loop Detection ............. 45

9. Security ............................................ 49

9.1 Confidentiality ..................................... 49

9.2 Authentication and Message Integrity ................ 50

10. RTP over Network and Transport Protocols ............ 51

11. Summary of Protocol Constants ....................... 51

11.1 RTCP packet types ................................... 52

11.2 SDES types .......................................... 52

12. RTP Profiles and Payload Format Specifications ...... 53

A. Algorithms .......................................... 56

A.1 RTP Data Header Validity Checks ..................... 59

A.2 RTCP Header Validity Checks ......................... 63

A.3 Determining the Number of RTP Packets Expected and

Lost ................................................ 63

A.4 Generating SDES RTCP Packets ........................ 64

A.5 Parsing RTCP SDES Packets ........................... 65

A.6 Generating a Random 32-bit Identifier ............... 66

A.7 Computing the RTCP Transmission Interval ............ 68

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A.8 Estimating the Interarrival Jitter .................. 71

B. Security Considerations ............................. 72

C. Addresses of Authors ................................ 72

D. Bibliography ........................................ 73

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 10). 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 multi-

participant 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,

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RFC 1889 RTP January 1996

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 [1]. That is, RTP is intended to be malleable

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 6.3.3.

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

RTP for a particular application will require one or more companion

documents (see Section 12):

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 TBD.

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 [2].

Several RTP applications, both experimental and commercial, have

already been implemented from draft specifications. These

applications include audio and video tools along with diagnostic

tools such as traffic monitors. Users of these tools number in the

thousands. However, the current Internet cannot yet support the full

potential demand for real-time services. High-bandwidth services

using RTP, such as video, can potentially seriously degrade the

quality of service of other network services. Thus, implementors

should take appropriate precautions to limit accidental bandwidth

usage. Application documentation should clearly outline the

limitations and possible operational impact of high-bandwidth real-

Schulzrinne, et al Standards Track [Page 4]

RFC 1889 RTP January 1996

time services on the Internet and other network services.

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 Internet-Draft draft-ietf-avt-profile

2.1 Simple Multicast Audio Conference

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

draft, 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,

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RFC 1889 RTP January 1996

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 6.5) when it leaves the

conference.

2.2 Audio and Video Conference

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

transmitted as separate RTP sessions 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 lower-

bandwidth 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

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RFC 1889 RTP January 1996

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.

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.

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 10).

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." [3] 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.

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RFC 1889 RTP January 1996

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 session: The association among a set of participants

communicating with RTP. For each participant, the session is

defined by a particular pair of destination transport addresses

(one network address plus a port pair for RTP and RTCP). The

destination transport address pair may be common for all

participants, as in the case of IP multicast, or may be

different for each, as in the case of individual unicast network

addresses plus a common port pair. In a multimedia session,

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

RTCP packets. The multiple RTP sessions are distinguished by

different port number pairs and/or different multicast

addresses.

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 6.4.1). 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 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).

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RFC 1889 RTP January 1996

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. These are called

third party monitors.

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 conference control application 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. For simple 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 [4].

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

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RFC 1889 RTP January 1996

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 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 [5]. 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.

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

Schulzrinne, et al Standards Track [Page 10]

RFC 1889 RTP January 1996

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

many padding octets should be ignored. 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 is 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.

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

specifies 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). An initial

set of default mappings for audio and video is specified in the

companion profile Internet-Draft draft-ietf-avt-profile, and

may be extended in future editions of the Assigned Numbers RFC

[6]. An RTP sender emits a single RTP payload type at any given

time; this field is not intended for multiplexing separate media

streams (see Section 5.2).

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 is random (unpredictable) to make known-plaintext attacks

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

encrypt, because the packets may flow through a translator that

does. Techniques for choosing unpredictable numbers are

discussed in [7].

timestamp: 32 bits

The timestamp reflects the sampling instant of the first octet

in the RTP data packet. The sampling instant must be derived

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RFC 1889 RTP January 1996

from a clock that increments monotonically and linearly in time

to allow synchronization and jitter calculations (see Section

6.3.1). 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 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 is random, as for the sequence

number. Several consecutive RTP packets may 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.)

SSRC: 32 bits

The SSRC field identifies the synchronization source. This

identifier is 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.

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 may be identified. CSRC identifiers are

Schulzrinne, et al Standards Track [Page 12]

RFC 1889 RTP January 1996

inserted by mixers, 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 [1]. In RTP, multiplexing is provided by the

destination transport address (network address and port number) which

define an 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. It is not intended that the audio and video be carried in a

single RTP session and demultiplexed based on the payload type or

SSRC fields. Interleaving packets with different payload types but

using the same SSRC would introduce several problems:

1. If one payload type were switched during a session, there

would be no general means to identify which of the old

values the new one replaced.

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 6.3) 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.

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RFC 1889 RTP January 1996

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 a 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 14]

RFC 1889 RTP January 1996

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 is 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 may 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. The feedback may be directly useful for control

of adaptive encodings [8,9], but experiments with IP

Schulzrinne, et al Standards Track [Page 15]

RFC 1889 RTP January 1996

multicasting have shown that it is also 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 6.3.

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

RTP source called the canonical name or CNAME, Section

6.4.1. 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 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.

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 are mandatory when RTP is used in the IP multicast

environment, and are recommended for all environments. RTP

application designers are advised to avoid mechanisms that can only

work in unicast mode and will not scale to larger numbers.

Schulzrinne, et al Standards Track [Page 16]

RFC 1889 RTP January 1996

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

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 always end on a 32-bit

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

part are included to make RTCP packets "stackable". Multiple RTCP

packets may 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:

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 should 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 should also include the SDES CNAME.

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

compound packet should be limited to increase the number of

constant bits in the first word and the probability of

successfully validating RTCP packets against misaddressed RTP

Schulzrinne, et al Standards Track [Page 17]

RFC 1889 RTP January 1996

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 recommended:

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

encrypted, it is prefixed by a random 32-bit quantity redrawn

for every compound packet transmitted.

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 nor

received, in which case an empty RR is 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. Other source description items may

optionally be included if required by a particular application,

subject to bandwidth constraints (see Section 6.2.2).

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.

It is advisable for translators and mixers to 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 maximum transmission unit (MTU) of

the network path, it may be segmented into multiple shorter compound

packets to be transmitted in separate packets of the underlying

protocol. Note that each of the compound packets must begin with an

SR or RR packet.

An implementation may ignore incoming RTCP packets with types unknown

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

Internet Assigned Numbers Authority (IANA).

Schulzrinne, et al Standards Track [Page 18]

RFC 1889 RTP January 1996

6.2 RTCP Transmission Interval

if encrypted: random 32-bit integer

|

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

|

| receiver reports chunk chunk

V item item item item

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

|R[SR|# sender #site#site][SDES|# CNAME PHONE |#CNAME LOC][BYE##why]

|R[ |# report # 1 # 2 ][ |# |# ][ ## ]

|R[ |# # # ][ |# |# ][ ## ]

|R[ |# # # ][ |# |# ][ ## ]

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

|<------------------ UDP packet (compound packet) --------------->|

#: SSRC/CSRC

Figure 1: Example of an RTCP compound packet

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 self-

limiting 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.

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, or it might just be a reasonable share. The

session bandwidth may be chosen based or 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. The session bandwidth

parameter is expected to be supplied by a session management

application when it invokes a media application, but media

applications may also 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.

Schulzrinne, et al Standards Track [Page 19]

RFC 1889 RTP January 1996

Bandwidth calculations for control and data traffic include lower-

layer 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.

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. It is suggested that the fraction

of the session bandwidth allocated to RTCP be fixed at 5%. While the

value of this and other constants in the interval calculation is 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.

The algorithm described in Appendix A.7 was designed to meet the

goals outlined above. 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:

o Senders are collectively allocated at least 1/4 of the control

traffic bandwidth 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.

o The calculated interval between RTCP packets is required to be

greater than a minimum of 5 seconds to avoid having bursts of

RTCP 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.

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 [10]. The first

RTCP packet sent after joining a session is also delayed by a

random variation of half the minimum RTCP interval in case the

application is started at multiple sites simultaneously, for

example as initiated by a session announcement.

Schulzrinne, et al Standards Track [Page 20]

RFC 1889 RTP January 1996

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

calculated, including all those received and sent, to

automatically adapt to changes in the amount of control

information carried.

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.

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 is created in

a table indexed by the SSRC or CSRC identifier (see Section 8.2) to

keep track of them. New entries may not be considered valid until

multiple packets carrying the new SSRC have been received (see

Appendix A.1). Entries may be deleted from the table when an RTCP BYE

packet with the corresponding SSRC identifier is received.

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 suggested). This provides some

robustness against packet loss. All sites must calculate roughly the

same value for the RTCP report interval in order for this timeout to

work properly.

Once a site has been validated, then if it is later marked inactive

the state for that site should still be retained and the site should

continue to be counted in the total number of sites sharing RTCP

bandwidth for a period long enough to span typical network

partitions. This is to avoid excessive traffic, when the partition

heals, due to an RTCP report interval that is too small. A timeout of

30 minutes is suggested. Note that this is still larger than 5 times

the largest value to which the RTCP report interval is expected to

usefully scale, about 2 to 5 minutes.

6.2.2 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

Schulzrinne, et al Standards Track [Page 21]

RFC 1889 RTP January 1996

bandwidth allocated to a single participant be used to carry the

additional information. Furthermore, it is not intended that all

SDES items should 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

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 might be sent in only one RTP session.

The other sessions would carry only the CNAME item.

6.3 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 may 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.

Schulzrinne, et al Standards Track [Page 22]

RFC 1889 RTP January 1996

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.

6.3.1 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

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

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

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

| SSRC of sender |

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

| NTP timestamp, most significant word | sender

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

| NTP timestamp, least significant word |

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

| RTP timestamp |

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

| sender's packet count |

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

| sender's octet count |

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

| SSRC_1 (SSRC of first source) | report

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

| fraction lost | cumulative number of packets lost | 1

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

| extended highest sequence number received |

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

| interarrival jitter |

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

| last SR (LSR) |

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

| delay since last SR (DLSR) |

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

| SSRC_2 (SSRC of second source) | report

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 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:

Schulzrinne, et al Standards Track [Page 23]

RFC 1889 RTP January 1996

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).

padding (P): 1 bit

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

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

control information. The last octet of the padding is a count of

how many padding octets should be ignored. Padding may be needed

by some encryption algorithms with fixed block sizes. In a

compound RTCP packet, padding should only be required on the

last individual packet because the compound packet is encrypted

as a whole.

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 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 may not be

known. A sender that can keep track of elapsed time but has no

notion of wallclock time may use the elapsed time since joining

Schulzrinne, et al Standards Track [Page 24]

RFC 1889 RTP January 1996

the session instead. This is assumed to be less than 68 years,

so the high bit will be zero. It is permissible to use the

sampling clock to estimate elapsed wallclock time. 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 is 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 is 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 is 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 do not carry over statistics when a source changes its SSRC

identifier due to a collision. These statistics are:

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

Schulzrinne, et al Standards Track [Page 25]

RFC 1889 RTP January 1996

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; 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.

Schulzrinne, et al Standards Track [Page 26]

RFC 1889 RTP January 1996

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

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

sampled.

The jitter calculation is prescribed here 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

[11]. A sample implementation is shown in Appendix A.8.

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 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 is illustrated

in Fig. 2.

This may be used as an approximate measure of distance to cluster

receivers, although some links have very asymmetric delays.

Schulzrinne, et al Standards Track [Page 27]

RFC 1889 RTP January 1996

6.3.2 RR: Receiver report RTCP packet

[10 Nov 1995 11:33:25.125] [10 Nov 1995 11:33:36.5]

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)

(3024992016.125 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 0x 6:2000 ( 6.125 s)

Figure 2: Example for round-trip time computation

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| RC | PT=RR=201 | length | header

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

| SSRC of packet sender |

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

| SSRC_1 (SSRC of first source) | report

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

| fraction lost | cumulative number of packets lost | 1

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

| extended highest sequence number received |

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

| interarrival jitter |

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

| last SR (LSR) |

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

| delay since last SR (DLSR) |

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

| SSRC_2 (SSRC of second source) | report

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

: ... : 2

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

| profile-specific extensions |

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

Schulzrinne, et al Standards Track [Page 28]

RFC 1889 RTP January 1996

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) is put at the head of a compound RTCP

packet when there is no data transmission or reception to report.

6.3.3 Extending the sender and receiver reports

A profile should define profile- or application-specific extensions

to the sender report and receiver if there is additional information

that should 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);

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.

If additional sender information is required, it should be included

first in the extension for sender reports, but would not be present

in receiver reports. If information about receivers is to be

included, that data may 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.

6.3.4 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

Schulzrinne, et al Standards Track [Page 29]

RFC 1889 RTP January 1996

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 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

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. Since the

interarrival jitter field is only a snapshot of the jitter at the

time of a report, it may be necessary to analyze a number of reports

from one receiver over time or from multiple receivers, e.g., within

a single network.

Schulzrinne, et al Standards Track [Page 30]

RFC 1889 RTP January 1996

6.4 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

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

|V=2|P| SC | PT=SDES=202 | length | header

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

| SSRC/CSRC_1 | chunk

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 1

| SDES items |

| ... |

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

| SSRC/CSRC_2 | chunk

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 2

| SDES items |

| ... |

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

The SDES packet is a three-level structure composed of a header and

zero or more chunks, each of 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 6.3.1).

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.

The text is encoded according to the UTF-2 encoding specified in

Annex F of ISO standard 10646 [12,13]. This encoding is also known as

UTF-8 or UTF-FSS. It is described in "File System Safe UCS

Transformation Format (FSS_UTF)", X/Open Preliminary Specification,

Document Number P316 and Unicode Technical Report #4. US-ASCII is a

subset of this encoding and requires no additional encoding. The

Schulzrinne, et al Standards Track [Page 31]

RFC 1889 RTP January 1996

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 multi-octet

encodings include null octets. The list of items in each chunk is

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,

and the remainder as needed to pad until the next 32-bit boundary. 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.

6.4.1 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 is required to provide the binding from the SSRC

identifier to an identifier for the source that remains

constant.

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.

Schulzrinne, et al Standards Track [Page 32]

RFC 1889 RTP January 1996

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 single-

user 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 [14], RFC 1035

[15] and Section 2.1 of RFC 1123 [16]; 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. 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 "doe@sleepy.megacorp.com" or "doe@192.0.2.89" for a

multi-user system. On a system with no user name, examples would be

"sleepy.megacorp.com" or "192.0.2.89".

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.

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 1597 [17]

may create network addresses that are not globally unique. This would

Schulzrinne, et al Standards Track [Page 33]

RFC 1889 RTP January 1996

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 [18].) 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.

6.4.2 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, Megacorp". 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.

6.4.3 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 822 [19], for

example, "John.Doe@megacorp.com". The EMAIL value is expected to

remain constant for the duration of a session.

6.4.4 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, "+1 908 555 1212" for a

number in the United States.

Schulzrinne, et al Standards Track [Page 34]

RFC 1889 RTP January 1996

6.4.5 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.

6.4.6 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.

6.4.7 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

Schulzrinne, et al Standards Track [Page 35]

RFC 1889 RTP January 1996

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.

6.4.8 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

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.

Schulzrinne, et al Standards Track [Page 36]

RFC 1889 RTP January 1996

6.5 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 |

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

: ... :

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

| length | reason for leaving ... (opt)

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

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 6.3.1).

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.

If a BYE packet is received by a mixer, the mixer forwards 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

is padded with null octets.

Schulzrinne, et al Standards Track [Page 37]

RFC 1889 RTP January 1996

6.6 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 the Internet Assigned Numbers Authority using an RTCP

packet type.

version (V), padding (P), length:

As described for the SR packet (see Section 6.3.1).

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.

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.

Schulzrinne, et al Standards Track [Page 38]

RFC 1889 RTP January 1996

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), multicast address or pair of

unicast addresses, and transport level destination port. (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.

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.

Schulzrinne, et al Standards Track [Page 39]

RFC 1889 RTP January 1996

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 will 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.

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

Schulzrinne, et al Standards Track [Page 40]

RFC 1889 RTP January 1996

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 multi-media 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 Figure 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 with

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

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

Schulzrinne, et al Standards Track [Page 41]

RFC 1889 RTP January 1996

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.

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.

Schulzrinne, et al Standards Track [Page 42]

RFC 1889 RTP January 1996

BYE: Translators forward BYE packets unchanged. Translators with

their own SSRC should generate BYE packets with that SSRC

identifier if they are about to cease forwarding packets.

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 generates its own SR packets

with sender information about the mixed data stream and sends

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 does not send these reception reports to the

other clouds and does 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.

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. 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.

Schulzrinne, et al Standards Track [Page 43]

RFC 1889 RTP January 1996

BYE: Mixers need to forward BYE packets. They should generate BYE

packets with their own SSRC identifiers if they are about to

cease forwarding packets.

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 Figure 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.

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

Schulzrinne, et al Standards Track [Page 44]

RFC 1889 RTP January 1996

independently pick the same value can be approximated for large N

[20] 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

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. 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 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

Schulzrinne, et al Standards Track [Page 45]

RFC 1889 RTP January 1996

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.

Consequently, if a source changes its source transport address, it

must also choose a new SSRC identifier to avoid being interpreted as

a looped source. 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. It 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 depends upon the source transport address being the

same for both RTP and RTCP packets from a source. The algorithm would

require modifications to support applications that don't meet this

constraint.

This algorithm requires keeping a table indexed by source identifiers

and containing the source transport address from which the identifier

was (first) received, along with other state for that source. Each

SSRC or CSRC identifier received in a data or control packet is

looked up in this table in order to process that data or control

information. For control packets, each element with its own SSRC,

for example an SDES chunk, requires a separate lookup. (The SSRC in a

reception report block is an exception.) If the SSRC or CSRC is not

Schulzrinne, et al Standards Track [Page 46]

RFC 1889 RTP January 1996

found, a new entry is created. These table entries are removed when

an RTCP BYE packet is received with the corresponding SSRC, or after

no packets have arrived for a relatively long time (see Section

6.2.1).

In order to track loops of the participant's own data packets, it is

also necessary to keep a separate list of source transport addresses

(not identifiers) that have been found to be conflicting. Note that

this should be a short list, usually empty. Each element in this list

stores the source address 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 the SSRC or CSRC identifier is not found in the source

identifier table:

THEN create a new entry storing the source transport address

and the SSRC or CSRC along with other state.

CONTINUE with normal processing.

(identifier is found in the table)

IF the source transport address from the packet matches

the one saved in the table entry for this identifier:

THEN CONTINUE with normal processing.

(an identifier collision or a loop is indicated)

IF the source identifier is not the participant's own:

THEN IF the source identifier is from an RTCP SDES chunk

containing a CNAME item that differs from the CNAME

in the table entry:

THEN (optionally) count a third-party collision.

ELSE (optionally) count a third-party loop.

ABORT processing of data packet or control element.

(a collision or loop of the participant's own data)

IF the source transport address is found in the list of

conflicting addresses:

THEN IF the source identifier is not from an RTCP SDES chunk

containing a CNAME item OR if that CNAME is the

participant's own:

Schulzrinne, et al Standards Track [Page 47]

RFC 1889 RTP January 1996

THEN (optionally) count occurrence of own traffic looped.

mark current time in conflicting address list entry.

ABORT processing of data packet or control element.

log occurrence of a collision.

create a new entry in the conflicting address list and

mark current time.

send an RTCP BYE packet with the old SSRC identifier.

choose a new identifier.

create a new entry in the source identifier table with the

old SSRC plus the source transport address from the packet

being processed.

CONTINUE with normal processing.

In this algorithm, packets from a newly conflicting source address

will be ignored and packets from the original source will be kept.

(If the original source was through a mixer and later the same source

is received directly, the receiver may be well advised to switch

unless other sources in the mix would be lost.) 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 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.

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 should be generated and the process

repeated.

A loop of data packets to a multicast destination can cause severe

network flooding. All mixers and translators are required to

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

might be attempted again periodically after a long, random time (on

the order of minutes).

Schulzrinne, et al Standards Track [Page 48]

RFC 1889 RTP January 1996

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

recently been specified for IP. Since the need for a confidentiality

service is well established in the initial audio and video

applications that are expected to use RTP, a confidentiality service

is defined in the next section for use with RTP and RTCP until lower

layer services are available. The overhead on the protocol for this

service is low, so the penalty will be minimal if this service is

obsoleted by lower layer services in the future.

Alternatively, other services, other implementations of services and

other algorithms may be defined for RTP in the future if warranted.

The selection presented here is meant to simplify implementation of

interoperable, secure applications and provide guidance to

implementors. 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 encryption of RTP or RTCP is desired, 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 is prepended to

the unit before encryption to deter known plaintext attacks. For RTP,

no prefix is required because the sequence number and timestamp

fields are initialized with random offsets.

For RTCP, it is allowed to split a compound RTCP packet into two

lower-layer 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 encrypted) to satisfy the requirement

that all compound RTCP packets begin with an SR or RR packet.

Schulzrinne, et al Standards Track [Page 49]

RFC 1889 RTP January 1996

UDP packet UDP packet

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

[32-bit ][ ][ # ] [ # sender # receiver]

[random ][ RR ][SDES # CNAME, ...] [ SR # report # report ]

[integer][(empty)][ # ] [ # # ]

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

encrypted not encrypted

#: SSRC

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.

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 [21], 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 [22].

Implementations that support encryption should always support the DES

algorithm in CBC mode as the default to maximize interoperability.

This method is chosen because it has been demonstrated to be easy and

practical to use in experimental audio and video tools in operation

on the Internet. Other encryption algorithms may be specified

dynamically for a session by non-RTP means.

As an alternative to encryption 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 should 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.

9.2 Authentication and Message Integrity

Authentication and message integrity are not defined in the current

specification of RTP 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 in the future.

Schulzrinne, et al Standards Track [Page 50]

RFC 1889 RTP January 1996

10. 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

uses an even port number and the corresponding RTCP stream uses the

next higher (odd) port number. If an application is supplied with an

odd number for use as the RTP port, it should replace this number

with the next lower (even) number.

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.

11. 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

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.

Schulzrinne, et al Standards Track [Page 51]

RFC 1889 RTP January 1996

11.1 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.

Other constants are assigned by IANA. Experimenters are encouraged to

register the numbers they need for experiments, and then unregister

those which prove to be unneeded.

11.2 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

Other constants are assigned by IANA. Experimenters are encouraged to

register the numbers they need for experiments, and then unregister

those which prove to be unneeded.

Schulzrinne, et al Standards Track [Page 52]

RFC 1889 RTP January 1996

12. 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 so there is no explicit indication of which

profile is in use. A profile for audio and video applications may be

found in the companion Internet-Draft draft-ietf-avt-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).

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).

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).

Schulzrinne, et al Standards Track [Page 53]

RFC 1889 RTP January 1996

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).

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 6.3.3).

SDES use: The profile may specify the relative priorities for RTCP

SDES items to be transmitted or excluded entirely (Section

6.2.2); an alternate syntax or semantics for the CNAME item

(Section 6.4.1); the format of the LOC item (Section 6.4.5); the

semantics and use of the NOTE item (Section 6.4.7); 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).

String-to-key mapping: A profile may specify how a user-provided

password or pass phrase is mapped into an encryption key.

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 10 may be specified.

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 10).

Schulzrinne, et al Standards Track [Page 54]

RFC 1889 RTP January 1996

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 the Internet Assigned

Numbers Authority and publishing their descriptions in an addendum to

the profile or in a payload format specification.

Schulzrinne, et al Standards Track [Page 55]

RFC 1889 RTP January 1996

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 big-

endian (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 (RFC XXXX)

*/

#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,

Schulzrinne, et al Standards Track [Page 56]

RFC 1889 RTP January 1996

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 */

u_int16 seq; /* 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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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 1889 RTP January 1996

/* 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 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, 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.

o The X bit must be zero if the profile does not specify that

the header extension mechanism may be used. Otherwise, the

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RFC 1889 RTP January 1996

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.

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 should be set to the number

of sequential packets required before declaring a source valid

(parameter MIN_SEQUENTIAL ) and s->max_seq initialized to seq-1 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.

Otherwise, the value zero is returned to indicate that the validation

failed, and the bad sequence number 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

Schulzrinne, et al Standards Track [Page 60]

RFC 1889 RTP January 1996

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 - 1;

s->max_seq = seq;

s->bad_seq = RTP_SEQ_MOD + 1;

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;

}

} 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) {

/*

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RFC 1889 RTP January 1996

* 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 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 could choose to discard all

data packets from a source until a valid RTCP packet has been

received from that source.

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.

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RFC 1889 RTP January 1996

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 can 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 only the last should possibly need

padding.

o The length fields of the individual RTCP packets must total 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);

if (r != end) {

/* something wrong with packet format */

}

A.3 Determining the Number of RTP Packets Expected and Lost

In order to compute packet loss rates, the number of 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

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RFC 1889 RTP January 1996

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 number is carried in 24 bits, it should be clamped at

0xffffff rather than wrap around to zero.

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.

A.4 Generating SDES RTCP Packets

This function builds one SDES chunk into buffer b composed of argc

items supplied in arrays type , value and length 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;

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RFC 1889 RTP January 1996

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;

}

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) {

Schulzrinne, et al Standards Track [Page 65]

RFC 1889 RTP January 1996

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 [23]. 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

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 [7].

Use of this or similar routine is suggested 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.

Schulzrinne, et al Standards Track [Page 66]

RFC 1889 RTP January 1996

/*

* 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);

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;

Schulzrinne, et al Standards Track [Page 67]

RFC 1889 RTP January 1996

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();

return md_32((char *)&s, sizeof(s));

} /* random32 */

A.7 Computing the RTCP Transmission Interval

The following function returns the time between transmissions of RTCP

packets, measured in seconds. It should be called after sending one

compound RTCP packet to calculate the delay until the next should be

sent. This function should also be called to calculate the delay

before sending the first RTCP packet upon startup rather than send

the packet immediately. This avoids any burst of RTCP packets if an

application is started at many sites simultaneously, for example as a

result of a session announcement.

The parameters have the following meaning:

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 should be 5% of the "session bandwidth"

parameter supplied to the application at startup.

senders: Number of active senders since sending last report, known

from construction of receiver reports for this RTCP packet.

Includes ourselves, if we also sent during this interval.

members: The estimated number of session members, including

ourselves. Incremented as we discover new session members from

the receipt of RTP or RTCP packets, and decremented as session

members leave (via RTCP BYE) or their state is timed out (30

minutes is recommended). On the first call, this parameter

should have the value 1.

Schulzrinne, et al Standards Track [Page 68]

RFC 1889 RTP January 1996

we_sent: Flag that is true if we have sent data during the last two

RTCP intervals. If the flag is true, the compound RTCP packet

just sent contained an SR packet.

packet_size: The size of the compound RTCP packet just sent, in

octets, including the network encapsulation (e.g., 28 octets for

UDP over IP).

avg_rtcp_size: Pointer to estimator for compound RTCP packet size;

initialized and updated by this function for the packet just

sent, and also updated by an identical line of code in the RTCP

receive routine for every RTCP packet received from other

participants in the session.

initial: Flag that is true for the first call upon startup to

calculate the time until the first report should be sent.

#include <math.h>

double rtcp_interval(int members,

int senders,

double rtcp_bw,

int we_sent,

int packet_size,

int *avg_rtcp_size,

int initial)

{

/*

* Minimum 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);

/*

* Gain (smoothing constant) for the low-pass filter that

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RFC 1889 RTP January 1996

* estimates the average RTCP packet size (see Cadzow reference).

*/

double const RTCP_SIZE_GAIN = (1./16.);

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. The average RTCP size is initialized

* to 128 octets which is conservative (it assumes everyone else

* is generating SRs instead of RRs: 20 IP + 8 UDP + 52 SR + 48

* SDES CNAME).

*/

if (initial) {

rtcp_min_time /= 2;

*avg_rtcp_size = 128;

}

/*

* If there were active senders, give them at least a minimum

* share of the RTCP bandwidth. Otherwise all participants share

* the RTCP bandwidth equally.

*/

n = members;

if (senders > 0 && 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;

}

}

/*

* Update the average size estimate by the size of the report

* packet we just sent.

*/

*avg_rtcp_size += (packet_size - *avg_rtcp_size)*RTCP_SIZE_GAIN;

/*

* The effective number of sites times the average packet size is

* the total number of octets sent when each site sends a report.

Schulzrinne, et al Standards Track [Page 70]

RFC 1889 RTP January 1996

* 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.

*/

return t * (drand48() + 0.5);

}

A.8 Estimating the Interarrival Jitter

The code fragments below implement the algorithm given in Section

6.3.1 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);

Schulzrinne, et al Standards Track [Page 71]

RFC 1889 RTP January 1996

In this case, the estimate is sampled for the reception report as:

rr->jitter = s->jitter >> 4;

B. 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 [24]. 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.

C. Authors' Addresses

Henning Schulzrinne

GMD Fokus

Hardenbergplatz 2

D-10623 Berlin

Germany

EMail: schulzrinne@fokus.gmd.de

Stephen L. Casner

Precept Software, Inc.

21580 Stevens Creek Boulevard, Suite 207

Cupertino, CA 95014

United States

EMail: casner@precept.com

Schulzrinne, et al Standards Track [Page 72]

RFC 1889 RTP January 1996

Ron Frederick

Xerox Palo Alto Research Center

3333 Coyote Hill Road

Palo Alto, CA 94304

United States

EMail: frederic@parc.xerox.com

Van Jacobson

MS 46a-1121

Lawrence Berkeley National Laboratory

Berkeley, CA 94720

United States

EMail: van@ee.lbl.gov

Acknowledgments

This memorandum is based on discussions within the IETF Audio/Video

Transport working group chaired by Stephen Casner. 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.

D. Bibliography

[1] D. D. Clark and D. L. Tennenhouse, "Architectural considerations

for a new generation of protocols," in SIGCOMM Symposium on

Communications Architectures and Protocols , (Philadelphia,

Pennsylvania), pp. 200--208, IEEE, Sept. 1990. Computer

Communications Review, Vol. 20(4), Sept. 1990.

[2] H. Schulzrinne, "Issues in designing a transport protocol for

audio and video conferences and other multiparticipant real-time

applications", Work in Progress.

[3] D. E. Comer, Internetworking with TCP/IP , vol. 1. Englewood

Cliffs, New Jersey: Prentice Hall, 1991.

[4] Postel, J., "Internet Protocol", STD 5, RFC 791, USC/Information

Sciences Institute, September 1981.

[5] Mills, D., "Network Time Protocol Version 3", RFC 1305, UDEL,

March 1992.

Schulzrinne, et al Standards Track [Page 73]

RFC 1889 RTP January 1996

[6] Reynolds, J., and J. Postel, "Assigned Numbers", STD 2, RFC 1700,

USC/Information Sciences Institute, October 1994.

[7] Eastlake, D., Crocker, S., and J. Schiller, "Randomness

Recommendations for Security", RFC 1750, DEC, Cybercash, MIT,

December 1994.

[8] J.-C. Bolot, T. Turletti, 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, Aug. 1994.

[9] I. Busse, B. Deffner, and H. Schulzrinne, "Dynamic QoS control of

multimedia applications based on RTP," Computer Communications ,

Jan. 1996.

[10] S. Floyd 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, Sept. 1993. also in [25].

[11] J. A. Cadzow, Foundations of digital signal processing and data

analysis New York, New York: Macmillan, 1987.

[12] International Standards Organization, "ISO/IEC DIS 10646-1:1993

information technology -- universal multiple-octet coded

character set (UCS) -- part I: Architecture and basic

multilingual plane," 1993.

[13] The Unicode Consortium, The Unicode Standard New York, New York:

Addison-Wesley, 1991.

[14] Mockapetris, P., "Domain Names - Concepts and Facilities", STD

13, RFC 1034, USC/Information Sciences Institute, November 1987.

[15] Mockapetris, P., "Domain Names - Implementation and

Specification", STD 13, RFC 1035, USC/Information Sciences

Institute, November 1987.

[16] Braden, R., "Requirements for Internet Hosts - Application and

Support", STD 3, RFC 1123, Internet Engineering Task Force,

October 1989.

[17] Rekhter, Y., Moskowitz, R., Karrenberg, D., and G. de Groot,

"Address Allocation for Private Internets", RFC 1597, T.J. Watson

Research Center, IBM Corp., Chrysler Corp., RIPE NCC, March 1994.

Schulzrinne, et al Standards Track [Page 74]

RFC 1889 RTP January 1996

[18] Lear, E., Fair, E., Crocker, D., and T. Kessler, "Network 10

Considered Harmful (Some Practices Shouldn't be Codified)", RFC

1627, Silicon Graphics, Inc., Apple Computer, Inc., Silicon

Graphics, Inc., July 1994.

[19] Crocker, D., "Standard for the Format of ARPA Internet Text

Messages", STD 11, RFC 822, UDEL, August 1982.

[20] W. Feller, An Introduction to Probability Theory and its

Applications, Volume 1 , vol. 1. New York, New York: John Wiley

and Sons, third ed., 1968.

[21] Balenson, D., "Privacy Enhancement for Internet Electronic Mail:

Part III: Algorithms, Modes, and Identifiers", RFC 1423, TIS, IAB

IRTF PSRG, IETF PEM WG, February 1993.

[22] V. L. Voydock and S. T. Kent, "Security mechanisms in high-level

network protocols," ACM Computing Surveys , vol. 15, pp. 135--

171, June 1983.

[23] Rivest, R., "The MD5 Message-Digest Algorithm", RFC 1321, MIT

Laboratory for Computer Science and RSA Data Security, Inc.,

April 1992.

[24] S. Stubblebine, "Security services for multimedia conferencing,"

in 16th National Computer Security Conference , (Baltimore,

Maryland), pp. 391--395, Sept. 1993.

[25] S. Floyd 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 75]

 
 
 
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