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Network Working Group Request for Comments: 3550 Obsoletes: 1889 Category: Standards Track |
H. Schulzrinne Columbia University S. Casner Packet Design R. Frederick Blue Coat Systems Inc. V. Jacobson Packet Design July 2003 |
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.
Copyright © The Internet Society (2003). All Rights Reserved.
This memorandum describes RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of-service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers.
Most of the text in this memorandum is identical to RFC 1889 which it obsoletes. There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously.
1. Introduction
1.1 Terminology
2. RTP Use Scenarios
2.1 Simple Multicast Audio Conference
2.2 Audio and Video Conference
2.3 Mixers and Translators
2.4 Layered Encodings
3. Definitions
4. Byte Order, Alignment, and Time Format
5. RTP Data Transfer Protocol
5.1 RTP Fixed Header Fields
5.2 Multiplexing RTP Sessions
5.3 Profile-Specific Modifications to the RTP Header
5.3.1 RTP Header Extension
6. RTP Control Protocol -- RTCP
6.1 RTCP Packet Format
6.2 RTCP Transmission Interval
6.2.1 Maintaining the Number of Session Members
6.3 RTCP Packet Send and Receive Rules
6.3.1 Computing the RTCP Transmission Interval
6.3.2 Initialization
6.3.3 Receiving an RTP or Non-BYE RTCP Packet
6.3.4 Receiving an RTCP BYE Packet
6.3.5 Timing Out an SSRC
6.3.6 Expiration of Transmission Timer
6.3.7 Transmitting a BYE Packet
6.3.8 Updating we_sent
6.3.9 Allocation of Source Description Bandwidth
6.4 Sender and Receiver Reports
6.4.1 SR: Sender Report RTCP Packet
6.4.2 RR: Receiver Report RTCP Packet
6.4.3 Extending the Sender and Receiver Reports
6.4.4 Analyzing Sender and Receiver Reports
6.5 SDES: Source Description RTCP Packet
6.5.1 CNAME: Canonical End-Point Identifier SDES Item . 46
6.5.2 NAME: User Name SDES Item
6.5.3 EMAIL: Electronic Mail Address SDES Item
6.5.4 PHONE: Phone Number SDES Item
6.5.5 LOC: Geographic User Location SDES Item
6.5.6 TOOL: Application or Tool Name SDES Item
6.5.7 NOTE: Notice/Status SDES Item
6.5.8 PRIV: Private Extensions SDES Item
6.6 BYE: Goodbye RTCP Packet
6.7 APP: Application-Defined RTCP Packet
7. RTP Translators and Mixers
7.1 General Description
7.2 RTCP Processing in Translators
7.3 RTCP Processing in Mixers
7.4 Cascaded Mixers
8. SSRC Identifier Allocation and Use
8.1 Probability of Collision
8.2 Collision Resolution and Loop Detection
8.3 Use with Layered Encodings
9. Security
9.1 Confidentiality
9.2 Authentication and Message Integrity
10. Congestion Control
11. RTP over Network and Transport Protocols
12. Summary of Protocol Constants
12.1 RTCP Packet Types
12.2 SDES Types
13. RTP Profiles and Payload Format Specifications
14. Security Considerations
15. IANA Considerations
16. Intellectual Property Rights Statement
17. Acknowledgments
Appendix A. Algorithms
Appendix A.1 RTP Data Header Validity Checks
Appendix A.2 RTCP Header Validity Checks
Appendix A.3 Determining Number of Packets Expected and Lost ... 83
Appendix A.4 Generating RTCP SDES Packets
Appendix A.5 Parsing RTCP SDES Packets
Appendix A.6 Generating a Random 32-bit Identifier
Appendix A.7 Computing the RTCP Transmission Interval
Appendix A.8 Estimating the Interarrival Jitter
Appendix B. Changes from RFC 1889
References
Normative References
Informative References
Authors' Addresses
Full Copyright Statement
This memorandum specifies the real-time transport protocol (RTP), which provides end-to-end delivery services for data with real-time characteristics, such as interactive audio and video. Those services include payload type identification, sequence numbering, timestamping and delivery monitoring. Applications typically run RTP on top of UDP to make use of its multiplexing and checksum services; both protocols contribute parts of the transport protocol functionality. However, RTP may be used with other suitable underlying network or transport protocols (see Section 11). RTP supports data transfer to multiple destinations using multicast distribution if provided by the underlying network.
Note that RTP itself does not provide any mechanism to ensure timely delivery or provide other quality-of-service guarantees, but relies on lower-layer services to do so. It does not guarantee delivery or prevent out-of-order delivery, nor does it assume that the underlying network is reliable and delivers packets in sequence. The sequence numbers included in RTP allow the receiver to reconstruct the sender's packet sequence, but sequence numbers might also be used to determine the proper location of a packet, for example in video decoding, without necessarily decoding packets in sequence.
While RTP is primarily designed to satisfy the needs of 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:
RTP represents a new style of protocol following the principles of application level framing and integrated layer processing proposed by Clark and Tennenhouse [10]. That is, RTP is intended to be malleable
to provide the information required by a particular application and will often be integrated into the application processing rather than being implemented as a separate layer. RTP is a protocol framework that is deliberately not complete. This document specifies those functions expected to be common across all the applications for which RTP would be appropriate. Unlike conventional protocols in which additional functions might be accommodated by making the protocol more general or by adding an option mechanism that would require parsing, RTP is intended to be tailored through modifications and/or additions to the headers as needed. Examples are given in Sections 5.3 and 6.4.3.
Therefore, in addition to this document, a complete specification of RTP for a particular application will require one or more companion documents (see Section 13):
A discussion of real-time services and algorithms for their implementation as well as background discussion on some of the RTP design decisions can be found in [11].
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14, RFC 2119 [2] and indicate requirement levels for compliant RTP implementations.
The following sections describe some aspects of the use of RTP. The examples were chosen to illustrate the basic operation of applications using RTP, not to limit what RTP may be used for. In these examples, RTP is carried on top of IP and UDP, and follows the conventions established by the profile for audio and video specified in the companion RFC 3551.
A working group of the IETF meets to discuss the latest protocol document, using the IP multicast services of the Internet for voice communications. Through some allocation mechanism the working group chair obtains a multicast group address and pair of ports. One port is used for audio data, and the other is used for control (RTCP) packets. This address and port information is distributed to the intended participants. If privacy is desired, the data and control packets may be encrypted as specified in Section 9.1, in which case an encryption key must also be generated and distributed. The exact details of these allocation and distribution mechanisms are beyond the scope of RTP.
The audio conferencing application used by each conference participant sends audio data in small chunks of, say, 20 ms duration. Each chunk of audio data is preceded by an RTP header; RTP header and data are in turn contained in a UDP packet. The RTP header indicates what type of audio encoding (such as PCM, ADPCM or LPC) is contained in each packet so that senders can change the encoding during a conference, for example, to accommodate a new participant that is connected through a low-bandwidth link or react to indications of network congestion.
The Internet, like other packet networks, occasionally loses and reorders packets and delays them by variable amounts of time. To cope with these impairments, the RTP header contains timing information and a sequence number that allow the receivers to reconstruct the timing produced by the source, so that in this example, chunks of audio are contiguously played out the speaker every 20 ms. This timing reconstruction is performed separately for each source of RTP packets in the conference. The sequence number can also be used by the receiver to estimate how many packets are being lost.
Since members of the working group join and leave during the conference, it is useful to know who is participating at any moment and how well they are receiving the audio data. For that purpose, each instance of the audio application in the conference periodically multicasts a reception report plus the name of its user on the RTCP (control) port. The reception report indicates how well the current speaker is being received and may be used to control adaptive encodings. In addition to the user name, other identifying information may also be included subject to control bandwidth limits. A site sends the RTCP BYE packet (Section 6.6) when it leaves the conference.
If both audio and video media are used in a conference, they are transmitted as separate RTP sessions. That is, separate RTP and RTCP packets are transmitted for each medium using two different UDP port pairs and/or multicast addresses. There is no direct coupling at the RTP level between the audio and video sessions, except that a user participating in both sessions should use the same distinguished (canonical) name in the RTCP packets for both so that the sessions can be associated.
One motivation for this separation is to allow some participants in the conference to receive only one medium if they choose. Further explanation is given in Section 5.2. Despite the separation, synchronized playback of a source's audio and video can be achieved using timing information carried in the RTCP packets for both sessions.
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 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.
Multimedia applications should be able to adjust the transmission rate to match the capacity of the receiver or to adapt to network congestion. Many implementations place the responsibility of rate- adaptivity at the source. This does not work well with multicast transmission because of the conflicting bandwidth requirements of heterogeneous receivers. The result is often a least-common denominator scenario, where the smallest pipe in the network mesh dictates the quality and fidelity of the overall live multimedia "broadcast".
Instead, responsibility for rate-adaptation can be placed at the receivers by combining a layered encoding with a layered transmission system. In the context of RTP over IP multicast, the source can stripe the progressive layers of a hierarchically represented signal across multiple RTP sessions each carried on its own multicast group. Receivers can then adapt to network heterogeneity and control their reception bandwidth by joining only the appropriate subset of the multicast groups.
Details of the use of RTP with layered encodings are given in Sections 6.3.9, 8.3 and 11.
RTP payload: The data transported by RTP in a packet, for example audio samples or compressed video data. The payload format and interpretation are beyond the scope of this document.
RTP packet: A data packet consisting of the fixed RTP header, a possibly empty list of contributing sources (see below), and the payload data. Some underlying protocols may require an encapsulation of the RTP packet to be defined. Typically one packet of the underlying protocol contains a single RTP packet, but several RTP packets MAY be contained if permitted by the encapsulation method (see Section 11).
RTCP packet: A control packet consisting of a fixed header part similar to that of RTP data packets, followed by structured elements that vary depending upon the RTCP packet type. The formats are defined in Section 6. Typically, multiple RTCP packets are sent together as a compound RTCP packet in a single packet of the underlying protocol; this is enabled by the length field in the fixed header of each RTCP packet.
Port: The "abstraction that transport protocols use to
distinguish among multiple destinations within a given host
computer. TCP/IP protocols identify ports using small positive
integers." [12] The transport selectors (TSEL) used by the OSI
transport layer are equivalent to ports. RTP depends upon the
lower-layer protocol to provide some mechanism such as ports to
multiplex the RTP and RTCP packets of a session.
Transport address: The combination of a network address and port that identifies a transport-level endpoint, for example an IP address and a UDP port. Packets are transmitted from a source transport address to a destination transport address.
RTP media type: An RTP media type is the collection of payload types which can be carried within a single RTP session. The RTP Profile assigns RTP media types to RTP payload types.
Multimedia session: A set of concurrent RTP sessions among a common group of participants. For example, a videoconference (which is a multimedia session) may contain an audio RTP session and a video RTP session.
RTP session: An association among a set of participants
communicating with RTP. A participant may be involved in multiple
RTP sessions at the same time. In a multimedia session, each
medium is typically carried in a separate RTP session with its own
RTCP packets unless the the encoding itself multiplexes multiple
media into a single data stream. A participant distinguishes
multiple RTP sessions by reception of different sessions using
different pairs of destination transport addresses, where a pair
of transport addresses comprises one network address plus a pair
of ports for RTP and RTCP. All participants in an RTP session may
share a common destination transport address pair, as in the case
of IP multicast, or the pairs may be different for each
participant, as in the case of individual unicast network
addresses and port pairs. In the unicast case, a participant may
receive from all other participants in the session using the same
pair of ports, or may use a distinct pair of ports for each.
The distinguishing feature of an RTP session is that each
maintains a full, separate space of SSRC identifiers (defined
next). The set of participants included in one RTP session
consists of those that can receive an SSRC identifier transmitted
by any one of the participants either in RTP as the SSRC or a CSRC
(also defined below) or in RTCP. For example, consider a three-
party conference implemented using unicast UDP with each
participant receiving from the other two on separate port pairs.
If each participant sends RTCP feedback about data received from
one other participant only back to that participant, then the
conference is composed of three separate point-to-point RTP
sessions. If each participant provides RTCP feedback about its
reception of one other participant to both of the other
participants, then the conference is composed of one multi-party
RTP session. The latter case simulates the behavior that would
occur with IP multicast communication among the three
participants.
The RTP framework allows the variations defined here, but a particular control protocol or application design will usually impose constraints on these variations.
Synchronization source (SSRC): The source of a stream of RTP packets, identified by a 32-bit numeric SSRC identifier carried in the RTP header so as not to be dependent upon the network address. All packets from a synchronization source form part of the same timing and sequence number space, so a receiver groups packets by synchronization source for playback. Examples of synchronization sources include the sender of a stream of packets derived from a signal source such as a microphone or a camera, or an RTP mixer (see below). A synchronization source may change its data format, e.g., audio encoding, over time. The SSRC identifier is a randomly chosen value meant to be globally unique within a particular RTP session (see Section 8). A participant need not use the same SSRC identifier for all the RTP sessions in a multimedia session; the binding of the SSRC identifiers is provided through RTCP (see Section 6.5.1). If a participant generates multiple streams in one RTP session, for example from separate video cameras, each MUST be identified as a different SSRC.
Contributing source (CSRC): A source of a stream of RTP packets that has contributed to the combined stream produced by an RTP mixer (see below). The mixer inserts a list of the SSRC identifiers of the sources that contributed to the generation of a particular packet into the RTP header of that packet. This list is called the CSRC list. An example application is audio conferencing where a mixer indicates all the talkers whose speech
was combined to produce the outgoing packet, allowing the receiver to indicate the current talker, even though all the audio packets contain the same SSRC identifier (that of the mixer).
End system: An application that generates the content to be sent in RTP packets and/or consumes the content of received RTP packets. An end system can act as one or more synchronization sources in a particular RTP session, but typically only one.
Mixer: An intermediate system that receives RTP packets from one or more sources, possibly changes the data format, combines the packets in some manner and then forwards a new RTP packet. Since the timing among multiple input sources will not generally be synchronized, the mixer will make timing adjustments among the streams and generate its own timing for the combined stream. Thus, all data packets originating from a mixer will be identified as having the mixer as their synchronization source.
Translator: An intermediate system that forwards RTP packets with their synchronization source identifier intact. Examples of translators include devices that convert encodings without mixing, replicators from multicast to unicast, and application-level filters in firewalls.
Monitor: An application that receives RTCP packets sent by participants in an RTP session, in particular the reception reports, and estimates the current quality of service for distribution monitoring, fault diagnosis and long-term statistics. The monitor function is likely to be built into the application(s) participating in the session, but may also be a separate application that does not otherwise participate and does not send or receive the RTP data packets (since they are on a separate port). These are called third-party monitors. It is also acceptable for a third-party monitor to receive the RTP data packets but not send RTCP packets or otherwise be counted in the session.
Non-RTP means: Protocols and mechanisms that may be needed in addition to RTP to provide a usable service. In particular, for multimedia conferences, a control protocol may distribute multicast addresses and keys for encryption, negotiate the encryption algorithm to be used, and define dynamic mappings between RTP payload type values and the payload formats they represent for formats that do not have a predefined payload type value. Examples of such protocols include the Session Initiation Protocol (SIP) (RFC 3261 [13]), ITU Recommendation H.323 [14] and applications using SDP (RFC 2327 [15]), such as RTSP (RFC 2326 [16]). For simple
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.
All integer fields are carried in network byte order, that is, most significant byte (octet) first. This byte order is commonly known as big-endian. The transmission order is described in detail in [3]. Unless otherwise noted, numeric constants are in decimal (base 10).
All header data is aligned to its natural length, i.e., 16-bit fields are aligned on even offsets, 32-bit fields are aligned at offsets divisible by four, etc. Octets designated as padding have the value zero.
Wallclock time (absolute date and time) is represented using the
timestamp format of the Network Time Protocol (NTP), which is in
seconds relative to 0h UTC on 1 January 1900 [4]. The full
resolution NTP timestamp is a 64-bit unsigned fixed-point number with
the integer part in the first 32 bits and the fractional part in the
last 32 bits. In some fields where a more compact representation is
appropriate, only the middle 32 bits are used; that is, the low 16
bits of the integer part and the high 16 bits of the fractional part.
The high 16 bits of the integer part must be determined
independently.
An implementation is not required to run the Network Time Protocol in order to use RTP. Other time sources, or none at all, may be used (see the description of the NTP timestamp field in Section 6.4.1). However, running NTP may be useful for synchronizing streams transmitted from separate hosts.
The NTP timestamp will wrap around to zero some time in the year 2036, but for RTP purposes, only differences between pairs of NTP timestamps are used. So long as the pairs of timestamps can be assumed to be within 68 years of each other, using modular arithmetic for subtractions and comparisons makes the wraparound irrelevant.
The RTP header has the following format:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC |M| PT | sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| contributing source (CSRC) identifiers |
| .... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The first twelve octets are present in every RTP packet, while the list of CSRC identifiers is present only when inserted by a mixer. The fields have the following meaning:
version (V): 2 bits
This field identifies the version of RTP. The version defined by
this specification is two (2). (The value 1 is used by the first
draft version of RTP and the value 0 is used by the protocol
initially implemented in the "vat" audio tool.)
padding (P): 1 bit
If the padding bit is set, the packet contains one or more
additional padding octets at the end which are not part of the
payload. The last octet of the padding contains a count of how
many padding octets should be ignored, including itself. Padding
may be needed by some encryption algorithms with fixed block sizes
or for carrying several RTP packets in a lower-layer protocol data
unit.
extension (X): 1 bit
If the extension bit is set, the fixed header MUST be followed by
exactly one header extension, with a format defined in Section
5.3.1.
CSRC count (CC): 4 bits
The CSRC count contains the number of CSRC identifiers that follow
the fixed header.
marker (M): 1 bit
The interpretation of the marker is defined by a profile. It is
intended to allow significant events such as frame boundaries to
be marked in the packet stream. A profile MAY define additional
marker bits or specify that there is no marker bit by changing the
number of bits in the payload type field (see Section 5.3).
payload type (PT): 7 bits
This field identifies the format of the RTP payload and determines
its interpretation by the application. A profile MAY specify a
default static mapping of payload type codes to payload formats.
Additional payload type codes MAY be defined dynamically through
non-RTP means (see Section 3). A set of default mappings for
audio and video is specified in the companion RFC 3551 [1]. An
RTP source MAY change the payload type during a session, but this
field SHOULD NOT be used for multiplexing separate media streams
(see Section 5.2).
A receiver MUST ignore packets with payload types that it does not understand.
sequence number: 16 bits
The sequence number increments by one for each RTP data packet
sent, and may be used by the receiver to detect packet loss and to
restore packet sequence. The initial value of the sequence number
SHOULD be random (unpredictable) to make known-plaintext attacks
on encryption more difficult, even if the source itself does not
encrypt according to the method in Section 9.1, because the
packets may flow through a translator that does. Techniques for
choosing unpredictable numbers are discussed in [17].
timestamp: 32 bits
The timestamp reflects the sampling instant of the first octet in
the RTP data packet. The sampling instant MUST be derived from a
clock that increments monotonically and linearly in time to allow
synchronization and jitter calculations (see Section 6.4.1). The
resolution of the clock MUST be sufficient for the desired
synchronization accuracy and for measuring packet arrival jitter
(one tick per video frame is typically not sufficient). The clock
frequency is dependent on the format of data carried as payload
and is specified statically in the profile or payload format
specification that defines the format, or MAY be specified
dynamically for payload formats defined through non-RTP means. If
RTP packets are generated periodically, the nominal sampling
instant as determined from the sampling clock is to be used, not a
reading of the system clock. As an example, for fixed-rate audio
the timestamp clock would likely increment by one for each
sampling period. If an audio application reads blocks covering
160 sampling periods from the input device, the timestamp would be increased by 160 for each such block, regardless of whether the block is transmitted in a packet or dropped as silent.
The initial value of the timestamp SHOULD be random, as for the sequence number. Several consecutive RTP packets will have equal timestamps if they are (logically) generated at once, e.g., belong to the same video frame. Consecutive RTP packets MAY contain timestamps that are not monotonic if the data is not transmitted in the order it was sampled, as in the case of MPEG interpolated video frames. (The sequence numbers of the packets as transmitted will still be monotonic.)
RTP timestamps from different media streams may advance at different rates and usually have independent, random offsets. Therefore, although these timestamps are sufficient to reconstruct the timing of a single stream, directly comparing RTP timestamps from different media is not effective for synchronization. Instead, for each medium the RTP timestamp is related to the sampling instant by pairing it with a timestamp from a reference clock (wallclock) that represents the time when the data corresponding to the RTP timestamp was sampled. The reference clock is shared by all media to be synchronized. The timestamp pairs are not transmitted in every data packet, but at a lower rate in RTCP SR packets as described in Section 6.4.
The sampling instant is chosen as the point of reference for the RTP timestamp because it is known to the transmitting endpoint and has a common definition for all media, independent of encoding delays or other processing. The purpose is to allow synchronized presentation of all media sampled at the same time.
Applications transmitting stored data rather than data sampled in real time typically use a virtual presentation timeline derived from wallclock time to determine when the next frame or other unit of each medium in the stored data should be presented. In this case, the RTP timestamp would reflect the presentation time for each unit. That is, the RTP timestamp for each unit would be related to the wallclock time at which the unit becomes current on the virtual presentation timeline. Actual presentation occurs some time later as determined by the receiver.
An example describing live audio narration of prerecorded video illustrates the significance of choosing the sampling instant as the reference point. In this scenario, the video would be presented locally for the narrator to view and would be simultaneously transmitted using RTP. The "sampling instant" of a video frame transmitted in RTP would be established by referencing
its timestamp to the wallclock time when that video frame was presented to the narrator. The sampling instant for the audio RTP packets containing the narrator's speech would be established by referencing the same wallclock time when the audio was sampled. The audio and video may even be transmitted by different hosts if the reference clocks on the two hosts are synchronized by some means such as NTP. A receiver can then synchronize presentation of the audio and video packets by relating their RTP timestamps using the timestamp pairs in RTCP SR packets.
SSRC: 32 bits
The SSRC field identifies the synchronization source. This
identifier SHOULD be chosen randomly, with the intent that no two
synchronization sources within the same RTP session will have the
same SSRC identifier. An example algorithm for generating a
random identifier is presented in Appendix A.6. Although the
probability of multiple sources choosing the same identifier is
low, all RTP implementations must be prepared to detect and
resolve collisions. Section 8 describes the probability of
collision along with a mechanism for resolving collisions and
detecting RTP-level forwarding loops based on the uniqueness of
the SSRC identifier. If a source changes its source transport
address, it must also choose a new SSRC identifier to avoid being
interpreted as a looped source (see Section 8.2).
CSRC list: 0 to 15 items, 32 bits each
The CSRC list identifies the contributing sources for the payload
contained in this packet. The number of identifiers is given by
the CC field. If there are more than 15 contributing sources,
only 15 can be identified. CSRC identifiers are inserted by
mixers (see Section 7.1), using the SSRC identifiers of
contributing sources. For example, for audio packets the SSRC
identifiers of all sources that were mixed together to create a
packet are listed, allowing correct talker indication at the
receiver.
For efficient protocol processing, the number of multiplexing points should be minimized, as described in the integrated layer processing design principle [10]. In RTP, multiplexing is provided by the destination transport address (network address and port number) which is different for each RTP session. For example, in a teleconference composed of audio and video media encoded separately, each medium SHOULD be carried in a separate RTP session with its own destination transport address.
Separate audio and video streams SHOULD NOT be carried in a single RTP session and demultiplexed based on the payload type or SSRC fields. Interleaving packets with different RTP media types but using the same SSRC would introduce several problems:
Using a different SSRC for each medium but sending them in the same RTP session would avoid the first three problems but not the last two.
On the other hand, multiplexing multiple related sources of the same medium in one RTP session using different SSRC values is the norm for multicast sessions. The problems listed above don't apply: an RTP mixer can combine multiple audio sources, for example, and the same treatment is applicable for all of them. It may also be appropriate to multiplex streams of the same medium using different SSRC values in other scenarios where the last two problems do not apply.
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.
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.
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.
Note that this header extension is intended only for limited use. Most potential uses of this mechanism would be better done another way, using the methods described in the previous section. For example, a profile-specific extension to the fixed header is less expensive to process because it is not conditional nor in a variable location. Additional information required for a particular payload format SHOULD NOT use this header extension, but SHOULD be carried in the payload section of the packet.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| defined by profile | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| header extension |
| .... |
If the X bit in the RTP header is one, a variable-length header extension MUST be appended to the RTP header, following the CSRC list if present. The header extension contains a 16-bit length field that counts the number of 32-bit words in the extension, excluding the four-octet extension header (therefore zero is a valid length). Only a single extension can be appended to the RTP data header. To allow multiple interoperating implementations to each experiment independently with different header extensions, or to allow a particular implementation to experiment with more than one type of header extension, the first 16 bits of the header extension are left open for distinguishing identifiers or parameters. The format of these 16 bits is to be defined by the profile specification under which the implementations are operating. This RTP specification does not define any header extensions itself.
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:
critical to get feedback from the receivers to diagnose faults in the distribution. Sending reception feedback reports to all participants allows one who is observing problems to evaluate whether those problems are local or global. With a distribution mechanism like IP multicast, it is also possible for an entity such as a network service provider who is not otherwise involved in the session to receive the feedback information and act as a third-party monitor to diagnose network problems. This feedback function is performed by the RTCP sender and receiver reports, described below in Section 6.4.
Functions 1-3 SHOULD be used in all environments, but particularly in the IP multicast environment. RTP application designers SHOULD avoid mechanisms that can only work in unicast mode and will not scale to larger numbers. Transmission of RTCP MAY be controlled separately for senders and receivers, as described in Section 6.2, for cases such as unidirectional links where feedback from receivers is not possible.
Non-normative note: In the multicast routing approach
called Source-Specific Multicast (SSM), there is only one sender
per "channel" (a source address, group address pair), and
receivers (except for the channel source) cannot use multicast to
communicate directly with other channel members. The
recommendations here accommodate SSM only through Section 6.2's
option of turning off receivers' RTCP entirely. Future work will
specify adaptation of RTCP for SSM so that feedback from receivers
can be maintained.
This specification defines several RTCP packet types to carry a variety of control information:
SR: Sender report, for transmission and reception statistics from
participants that are active senders
RR: Receiver report, for reception statistics from participants
that are not active senders and in combination with SR for
active senders reporting on more than 31 sources
SDES: Source description items, including CNAME
BYE: Indicates end of participation
APP: Application-specific functions
Each RTCP packet begins with a fixed part similar to that of RTP data packets, followed by structured elements that MAY be of variable length according to the packet type but MUST end on a 32-bit boundary. The alignment requirement and a length field in the fixed part of each packet are included to make RTCP packets "stackable". Multiple RTCP packets can be concatenated without any intervening separators to form a compound RTCP packet that is sent in a single packet of the lower layer protocol, for example UDP. There is no explicit count of individual RTCP packets in the compound packet since the lower layer protocols are expected to provide an overall length to determine the end of the compound packet.
Each individual RTCP packet in the compound packet may be processed independently with no requirements upon the order or combination of packets. However, in order to perform the functions of the protocol, the following constraints are imposed:
Thus, all RTCP packets MUST be sent in a compound packet of at least two individual packets, with the following format:
Encryption prefix: If and only if the compound packet is to be encrypted according to the method in Section 9.1, it MUST be prefixed by a random 32-bit quantity redrawn for every compound packet transmitted. If padding is required for the encryption, it MUST be added to the last packet of the compound packet.
SR or RR: The first RTCP packet in the compound packet MUST always be a report packet to facilitate header validation as described in Appendix A.2. This is true even if no data has been sent or received, in which case an empty RR MUST be sent, and even if the only other RTCP packet in the compound packet is a BYE.
Additional RRs: If the number of sources for which reception statistics are being reported exceeds 31, the number that will fit into one SR or RR packet, then additional RR packets SHOULD follow the initial report packet.
SDES: An SDES packet containing a CNAME item MUST be included in each compound RTCP packet, except as noted in Section 9.1. Other source description items MAY optionally be included if required by a particular application, subject to bandwidth constraints (see Section 6.3.9).
BYE or APP: Other RTCP packet types, including those yet to be defined, MAY follow in any order, except that BYE SHOULD be the last packet sent with a given SSRC/CSRC. Packet types MAY appear more than once.
An individual RTP participant SHOULD send only one compound RTCP packet per report interval in order for the RTCP bandwidth per participant to be estimated correctly (see Section 6.2), except when the compound RTCP packet is split for partial encryption as described in Section 9.1. If there are too many sources to fit all the necessary RR packets into one compound RTCP packet without exceeding the maximum transmission unit (MTU) of the network path, then only the subset that will fit into one MTU SHOULD be included in each interval. The subsets SHOULD be selected round-robin across multiple intervals so that all sources are reported.
It is RECOMMENDED that translators and mixers combine individual RTCP packets from the multiple sources they are forwarding into one compound packet whenever feasible in order to amortize the packet overhead (see Section 7). An example RTCP compound packet as might be produced by a mixer is shown in Fig. 1. If the overall length of a compound packet would exceed the MTU of the network path, it SHOULD be segmented into multiple shorter compound packets to be transmitted in separate packets of the underlying protocol. This does not impair the RTCP bandwidth estimation because each compound packet represents at least one distinct participant. Note that each of the compound packets MUST begin with an SR or RR packet.
An implementation SHOULD ignore incoming RTCP packets with types unknown to it. Additional RTCP packet types may be registered with the Internet Assigned Numbers Authority (IANA) as described in Section 15.
if encrypted: random 32-bit integer
| |[--------- packet --------][---------- packet ----------][-packet-] | | receiver chunk chunk V reports item item item item -------------------------------------------------------------------- R[SR #sendinfo #site1#site2][SDES #CNAME PHONE #CNAME LOC][BYE##why] -------------------------------------------------------------------- | | |<----------------------- compound packet ----------------------->| |<-------------------------- UDP packet ------------------------->| #: SSRC/CSRC identifier
Figure 1: Example of an RTCP compound packet
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 by dynamically calculating the interval between RTCP packet transmissions.
For each session, it is assumed that the data traffic is subject to an aggregate limit called the "session bandwidth" to be divided among the participants. This bandwidth might be reserved and the limit enforced by the network. If there is no reservation, there may be other constraints, depending on the environment, that establish the "reasonable" maximum for the session to use, and that would be the session bandwidth. The session bandwidth may be chosen based on some cost or a priori knowledge of the available network bandwidth for the session. It is somewhat independent of the media encoding, but the encoding choice may be limited by the session bandwidth. Often, the session bandwidth is the sum of the nominal bandwidths of the senders expected to be concurrently active. For teleconference audio, this number would typically be one sender's bandwidth. For layered encodings, each layer is a separate RTP session with its own session bandwidth parameter.
The session bandwidth parameter is expected to be supplied by a session management application when it invokes a media application, but media applications MAY set a default based on the single-sender data bandwidth for the encoding selected for the session. The application MAY also enforce bandwidth limits based on multicast scope rules or other criteria. All participants MUST use the same value for the session bandwidth so that the same RTCP interval will be calculated.
Bandwidth calculations for control and data traffic include 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. The control traffic bandwidth is in addition to the session bandwidth for the data traffic. It is RECOMMENDED that the fraction of the session bandwidth added for RTCP be fixed at 5%. It is also RECOMMENDED that 1/4 of the RTCP bandwidth be dedicated to participants that are sending data so that in sessions with a large number of receivers but a small number of senders, newly joining participants will more quickly receive the CNAME for the sending sites. When the proportion of senders is greater than 1/4 of the participants, the senders get their proportion of the full RTCP bandwidth. While the values of these and other constants in the interval calculation are not critical, all participants in the session MUST use the same values so the same interval will be calculated. Therefore, these constants SHOULD be fixed for a particular profile.
A profile MAY specify that the control traffic bandwidth may be a separate parameter of the session rather than a strict percentage of the session bandwidth. Using a separate parameter allows rate- adaptive applications to set an RTCP bandwidth consistent with a "typical" data bandwidth that is lower than the maximum bandwidth specified by the session bandwidth parameter.
The profile MAY further specify that the control traffic bandwidth may be divided into two separate session parameters for those participants which are active data senders and those which are not; let us call the parameters S and R. Following the recommendation that 1/4 of the RTCP bandwidth be dedicated to data senders, the RECOMMENDED default values for these two parameters would be 1.25% and 3.75%, respectively. When the proportion of senders is greater than S/(S+R) of the participants, the senders get their proportion of the sum of these parameters. Using two parameters allows RTCP reception reports to be turned off entirely for a particular session by setting the RTCP bandwidth for non-data-senders to zero while keeping the RTCP bandwidth for data senders non-zero so that sender reports can still be sent for inter-media synchronization. Turning off RTCP reception reports is NOT RECOMMENDED because they are needed for the functions listed at the beginning of Section 6, particularly reception quality feedback and congestion control. However, doing so may be appropriate for systems operating on unidirectional links or for sessions that don't require feedback on the quality of reception or liveness of receivers and that have other means to avoid congestion.
The calculated interval between transmissions of compound RTCP packets SHOULD also have a lower bound to avoid having bursts of packets exceed the allowed bandwidth when the number of participants is small and the traffic isn't smoothed according to the law of large numbers. It also keeps the report interval from becoming too small during transient outages like a network partition such that adaptation is delayed when the partition heals. At application startup, a delay SHOULD be imposed before the first compound RTCP packet is sent to allow time for RTCP packets to be received from other participants so the report interval will converge to the correct value more quickly. This delay MAY be set to half the minimum interval to allow quicker notification that the new participant is present. The RECOMMENDED value for a fixed minimum interval is 5 seconds.
An implementation MAY scale the minimum RTCP interval to a smaller value inversely proportional to the session bandwidth parameter with the following limitations:
The algorithm described in Section 6.3 and Appendix A.7 was designed to meet the goals outlined in this section. It calculates the interval between sending compound RTCP packets to divide the allowed control traffic bandwidth among the participants. This allows an application to provide fast response for small sessions where, for example, identification of all participants is important, yet automatically adapt to large sessions. The algorithm incorporates the following characteristics:
This algorithm may be used for sessions in which all participants are allowed to send. In that case, the session bandwidth parameter is the product of the individual sender's bandwidth times the number of participants, and the RTCP bandwidth is 5% of that.
Details of the algorithm's operation are given in the sections that follow. Appendix A.7 gives an example implementation.
Calculation of the RTCP packet interval depends upon an estimate of the number of sites participating in the session. New sites are added to the count when they are heard, and an entry for each SHOULD be created in a table indexed by the SSRC or CSRC identifier (see Section 8.2) to keep track of them. New entries MAY be considered not valid until multiple packets carrying the new SSRC have been received (see Appendix A.1), or until an SDES RTCP packet containing a CNAME for that SSRC has been received. Entries MAY be deleted from the table when an RTCP BYE packet with the corresponding SSRC identifier is received, except that some straggler data packets might arrive after the BYE and cause the entry to be recreated. Instead, the entry SHOULD be marked as having received a BYE and then deleted after an appropriate delay.
A participant MAY mark another site inactive, or delete it if not yet valid, if no RTP or RTCP packet has been received for a small number of RTCP report intervals (5 is RECOMMENDED). This provides some robustness against packet loss. All sites must have the same value for this multiplier and must calculate roughly the same value for the RTCP report interval in order for this timeout to work properly. Therefore, this multiplier SHOULD be fixed for a particular profile.
For sessions with a very large number of participants, it may be impractical to maintain a table to store the SSRC identifier and state information for all of them. An implementation MAY use SSRC sampling, as described in [21], to reduce the storage requirements. An implementation MAY use any other algorithm with similar performance. A key requirement is that any algorithm considered SHOULD NOT substantially underestimate the group size, although it MAY overestimate.
The rules for how to send, and what to do when receiving an RTCP packet are outlined here. An implementation that allows operation in a multicast environment or a multipoint unicast environment MUST meet the requirements in Section 6.2. Such an implementation MAY use the algorithm defined in this section to meet those requirements, or MAY use some other algorithm so long as it provides equivalent or better performance. An implementation which is constrained to two-party unicast operation SHOULD still use randomization of the RTCP transmission interval to avoid unintended synchronization of multiple instances operating in the same environment, but MAY omit the "timer reconsideration" and "reverse reconsideration" algorithms in Sections 6.3.3, 6.3.6 and 6.3.7.
To execute these rules, a session participant must maintain several pieces of state:
tp: the last time an RTCP packet was transmitted;
tc: the current time;
tn: the next scheduled transmission time of an RTCP packet;
pmembers: the estimated number of session members at the time tn was last recomputed;
members: the most current estimate for the number of session members;
senders: the most current estimate for the number of senders in the session;
rtcp_bw: The target RTCP bandwidth, i.e., the total bandwidth that will be used for RTCP packets by all members of this session, in octets per second. This will be a specified fraction of the "session bandwidth" parameter supplied to the application at startup.
we_sent: Flag that is true if the application has sent data since the 2nd previous RTCP report was transmitted.
avg_rtcp_size: The average compound RTCP packet size, in octets, over all RTCP packets sent and received by this participant. The size includes lower-layer transport and network protocol headers (e.g., UDP and IP) as explained in Section 6.2.
initial: Flag that is true if the application has not yet sent an RTCP packet.
Many of these rules make use of the "calculated interval" between packet transmissions. This interval is described in the following section.
To maintain scalability, the average interval between packets from a session participant should scale with the group size. This interval is called the calculated interval. It is obtained by combining a number of the pieces of state described above. The calculated interval T is then determined as follows:
This procedure results in an interval which is random, but which, on average, gives at least 25% of the RTCP bandwidth to senders and the rest to receivers. If the senders constitute more than one quarter of the membership, this procedure splits the bandwidth equally among all participants, on average.
Upon joining the session, the participant initializes tp to 0, tc to 0, senders to 0, pmembers to 1, members to 1, we_sent to false, rtcp_bw to the specified fraction of the session bandwidth, initial to true, and avg_rtcp_size to the probable size of the first RTCP packet that the application will later construct. The calculated interval T is then computed, and the first packet is scheduled for
time tn = T. This means that a transmission timer is set which expires at time T. Note that an application MAY use any desired approach for implementing this timer.
The participant adds its own SSRC to the member table.
When an RTP or RTCP packet is received from a participant whose SSRC is not in the member table, the SSRC is added to the table, and the value for members is updated once the participant has been validated as described in Section 6.2.1. The same processing occurs for each CSRC in a validated RTP packet.
When an RTP packet is received from a participant whose SSRC is not in the sender table, the SSRC is added to the table, and the value for senders is updated.
For each compound RTCP packet received, the value of avg_rtcp_size is updated:
avg_rtcp_size = (1/16) * packet_size + (15/16) * avg_rtcp_size
where packet_size is the size of the RTCP packet just received.
Except as described in Section 6.3.7 for the case when an RTCP BYE is to be transmitted, if the received packet is an RTCP BYE packet, the SSRC is checked against the member table. If present, the entry is removed from the table, and the value for members is updated. The SSRC is then checked against the sender table. If present, the entry is removed from the table, and the value for senders is updated.
Furthermore, to make the transmission rate of RTCP packets more adaptive to changes in group membership, the following "reverse reconsideration" algorithm SHOULD be executed when a BYE packet is received that reduces members to a value less than pmembers:
tn = tc + (members/pmembers) * (tn - tc)
tp = tc - (members/pmembers) * (tc - tp).
This algorithm does not prevent the group size estimate from incorrectly dropping to zero for a short time due to premature timeouts when most participants of a large session leave at once but some remain. The algorithm does make the estimate return to the correct value more rapidly. This situation is unusual enough and the consequences are sufficiently harmless that this problem is deemed only a secondary concern.
At occasional intervals, the participant MUST check to see if any of the other participants time out. To do this, the participant computes the deterministic (without the randomization factor) calculated interval Td for a receiver, that is, with we_sent false. Any other session member who has not sent an RTP or RTCP packet since time tc - MTd (M is the timeout multiplier, and defaults to 5) is timed out. This means that its SSRC is removed from the member list, and members is updated. A similar check is performed on the sender list. Any member on the sender list who has not sent an RTP packet since time tc - 2T (within the last two RTCP report intervals) is removed from the sender list, and senders is updated.
If any members time out, the reverse reconsideration algorithm described in Section 6.3.4 SHOULD be performed.
The participant MUST perform this check at least once per RTCP transmission interval.
When the packet transmission timer expires, the participant performs the following operations:
If an RTCP packet is transmitted, the value of initial is set to FALSE. Furthermore, the value of avg_rtcp_size is updated:
avg_rtcp_size = (1/16) * packet_size + (15/16) * avg_rtcp_size
where packet_size is the size of the RTCP packet just transmitted.
When a participant wishes to leave a session, a BYE packet is transmitted to inform the other participants of the event. In order to avoid a flood of BYE packets when many participants leave the system, a participant MUST execute the following algorithm if the number of members is more than 50 when the participant chooses to leave. This algorithm usurps the normal role of the members variable to count BYE packets instead:
scheduled for time tn = tc + T.
This allows BYE packets to be sent right away, yet controls their total bandwidth usage. In the worst case, this could cause RTCP control packets to use twice the bandwidth as normal (10%) -- 5% for non-BYE RTCP packets and 5% for BYE.
A participant that does not want to wait for the above mechanism to allow transmission of a BYE packet MAY leave the group without sending a BYE at all. That participant will eventually be timed out by the other group members.
If the group size estimate members is less than 50 when the participant decides to leave, the participant MAY send a BYE packet immediately. Alternatively, the participant MAY choose to execute the above BYE backoff algorithm.
In either case, a participant which never sent an RTP or RTCP packet MUST NOT send a BYE packet when they leave the group.
The variable we_sent contains true if the participant has sent an RTP packet recently, false otherwise. This determination is made by using the same mechanisms as for managing the set of other participants listed in the senders table. If the participant sends an RTP packet when we_sent is false, it adds itself to the sender table and sets we_sent to true. The reverse reconsideration algorithm described in Section 6.3.4 SHOULD be performed to possibly reduce the delay before sending an SR packet. Every time another RTP packet is sent, the time of transmission of that packet is maintained in the table. The normal sender timeout algorithm is then applied to the participant -- if an RTP packet has not been transmitted since time tc - 2T, the participant removes itself from the sender table, decrements the sender count, and sets we_sent to false.
This specification defines several source description (SDES) items in addition to the mandatory CNAME item, such as NAME (personal name) and EMAIL (email address). It also provides a means to define new application-specific RTCP packet types. Applications should exercise caution in allocating control bandwidth to this additional information because it will slow down the rate at which reception reports and CNAME are sent, thus impairing the performance of the protocol. It is RECOMMENDED that no more than 20% of the RTCP bandwidth allocated to a single participant be used to carry the additional information. Furthermore, it is not intended that all SDES items will be included in every application. Those that are included SHOULD be assigned a fraction of the bandwidth according to their utility. Rather than estimate these fractions dynamically, it is recommended that the percentages be translated statically into report interval counts based on the typical length of an item.
For example, an application may be designed to send only CNAME, NAME and EMAIL and not any others. NAME might be given much higher priority than EMAIL because the NAME would be displayed continuously in the application's user interface, whereas EMAIL would be displayed only when requested. At every RTCP interval, an RR packet and an SDES packet with the CNAME item would be sent. For a small session
operating at the minimum interval, that would be every 5 seconds on the average. Every third interval (15 seconds), one extra item would be included in the SDES packet. Seven out of eight times this would be the NAME item, and every eighth time (2 minutes) it would be the EMAIL item.
When multiple applications operate in concert using cross-application binding through a common CNAME for each participant, for example in a multimedia conference composed of an RTP session for each medium, the additional SDES information MAY be sent in only one RTP session. The other sessions would carry only the CNAME item. In particular, this approach should be applied to the multiple sessions of a layered encoding scheme (see Section 2.4).
RTP receivers provide reception quality feedback using RTCP report packets which may take one of two forms depending upon whether or not the receiver is also a sender. The only difference between the sender report (SR) and receiver report (RR) forms, besides the packet type code, is that the sender report includes a 20-byte sender information section for use by active senders. The SR is issued if a site has sent any data packets during the interval since issuing the last report or the previous one, otherwise the RR is issued.
Both the SR and RR forms include zero or more reception report blocks, one for each of the synchronization sources from which this receiver has received RTP data packets since the last report. Reports are not issued for contributing sources listed in the CSRC list. Each reception report block provides statistics about the data received from the particular source indicated in that block. Since a maximum of 31 reception report blocks will fit in an SR or RR packet, additional RR packets SHOULD be stacked after the initial SR or RR packet as needed to contain the reception reports for all sources heard during the interval since the last report. If there are too many sources to fit all the necessary RR packets into one compound RTCP packet without exceeding the MTU of the network path, then only the subset that will fit into one MTU SHOULD be included in each interval. The subsets SHOULD be selected round-robin across multiple intervals so that all sources are reported.
The next sections define the formats of the two reports, how they may be extended in a profile-specific manner if an application requires additional feedback information, and how the reports may be used. Details of reception reporting by translators and mixers is given in Section 7.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of sender |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| NTP timestamp, least significant word |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTP timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| sender's packet count |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| sender's octet count |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
1 | fraction lost | cumulative number of packets lost |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| extended highest sequence number received |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| interarrival jitter |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| last SR (LSR) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| delay since last SR (DLSR) |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
2 : ... :
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| profile-specific extensions |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The sender report packet consists of three sections, possibly followed by a fourth profile-specific extension section if defined. The first section, the header, is 8 octets long. The fields have the following meaning:
version (V): 2 bits
Identifies the version of RTP, which is the same in RTCP packets
as in RTP data packets. The version defined by this specification
is two (2).
padding (P): 1 bit
If the padding bit is set, this individual RTCP packet contains
some additional padding octets at the end which are not part of
the control information but are included in the length field. The
last octet of the padding is a count of how many padding octets
should be ignored, including itself (it will be a multiple of
four). Padding may be needed by some encryption algorithms with
fixed block sizes. In a compound RTCP packet, padding is only
required on one individual packet because the compound packet is
encrypted as a whole for the method in Section 9.1. Thus, padding
MUST only be added to the last individual packet, and if padding
is added to that packet, the padding bit MUST be set only on that
packet. This convention aids the header validity checks described
in Appendix A.2 and allows detection of packets from some early
implementations that incorrectly set the padding bit on the first
individual packet and add padding to the last individual packet.
reception report count (RC): 5 bits
The number of reception report blocks contained in this packet. A
value of zero is valid.
packet type (PT): 8 bits
Contains the constant 200 to identify this as an RTCP SR packet.
length: 16 bits
The length of this RTCP packet in 32-bit words minus one,
including the header and any padding. (The offset of one makes
zero a valid length and avoids a possible infinite loop in
scanning a compound RTCP packet, while counting 32-bit words
avoids a validity check for a multiple of 4.)
SSRC: 32 bits
The synchronization source identifier for the originator of this
SR packet.
The second section, the sender information, is 20 octets long and is present in every sender report packet. It summarizes the data transmissions from this sender. The fields have the following meaning:
NTP timestamp: 64 bits
Indicates the wallclock time (see Section 4) when this report was
sent so that it may be used in combination with timestamps
returned in reception reports from other receivers to measure
round-trip propagation to those receivers. Receivers should
expect that the measurement accuracy of the timestamp may be
limited to far less than the resolution of the NTP timestamp. The
measurement uncertainty of the timestamp is not indicated as it
may not be known. On a system that has no notion of wallclock
time but does have some system-specific clock such as "system
uptime", a sender MAY use that clock as a reference to calculate
relative NTP timestamps. It is important to choose a commonly
used clock so that if separate implementations are used to produce
the individual streams of a multimedia session, all
implementations will use the same clock. Until the year 2036,
relative and absolute timestamps will differ in the high bit so
(invalid) comparisons will show a large difference; by then one
hopes relative timestamps will no longer be needed. A sender that
has no notion of wallclock or elapsed time MAY set the NTP
timestamp to zero.
RTP timestamp: 32 bits
Corresponds to the same time as the NTP timestamp (above), but in
the same units and with the same random offset as the RTP
timestamps in data packets. This correspondence may be used for
intra- and inter-media synchronization for sources whose NTP
timestamps are synchronized, and may be used by media-independent
receivers to estimate the nominal RTP clock frequency. Note that
in most cases this timestamp will not be equal to the RTP
timestamp in any adjacent data packet. Rather, it MUST be
calculated from the corresponding NTP timestamp using the
relationship between the RTP timestamp counter and real time as
maintained by periodically checking the wallclock time at a
sampling instant.
sender's packet count: 32 bits
The total number of RTP data packets transmitted by the sender
since starting transmission up until the time this SR packet was
generated. The count SHOULD be reset if the sender changes its
SSRC identifier.
sender's octet count: 32 bits
The total number of payload octets (i.e., not including header or
padding) transmitted in RTP data packets by the sender since
starting transmission up until the time this SR packet was
generated. The count SHOULD be reset if the sender changes its
SSRC identifier. This field can be used to estimate the average
payload data rate.
The third section contains zero or more reception report blocks depending on the number of other sources heard by this sender since the last report. Each reception report block conveys statistics on the reception of RTP packets from a single synchronization source. Receivers SHOULD NOT carry over statistics when a source changes its SSRC identifier due to a collision. These statistics are:
SSRC_n (source identifier): 32 bits
The SSRC identifier of the source to which the information in this
reception report block pertains.
fraction lost: 8 bits
The fraction of RTP data packets from source SSRC_n lost since the
previous SR or RR packet was sent, expressed as a fixed point
number with the binary point at the left edge of the field. (That
is equivalent to taking the integer part after multiplying the
loss fraction by 256.) This fraction is defined to be the number
of packets lost divided by the number of packets expected, as
defined in the next paragraph. An implementation is shown in
Appendix A.3. If the loss is negative due to duplicates, the
fraction lost is set to zero. Note that a receiver cannot tell
whether any packets were lost after the last one received, and
that there will be no reception report block issued for a source
if all packets from that source sent during the last reporting
interval have been lost.
cumulative number of packets lost: 24 bits
The total number of RTP data packets from source SSRC_n that have
been lost since the beginning of reception. This number is
defined to be the number of packets expected less the number of
packets actually received, where the number of packets received
includes any which are late or duplicates. Thus, packets that
arrive late are not counted as lost, and the loss may be negative
if there are duplicates. The number of packets expected is
defined to be the extended last sequence number received, as
defined next, less the initial sequence number received. This may
be calculated as shown in Appendix A.3.
extended highest sequence number received: 32 bits
The low 16 bits contain the highest sequence number received in an
RTP data packet from source SSRC_n, and the most significant 16
bits extend that sequence number with the corresponding count of
sequence number cycles, which may be maintained according to the
algorithm in Appendix A.1. Note that different receivers within
the same session will generate different extensions to the
sequence number if their start times differ significantly.
interarrival jitter: 32 bits
An estimate of the statistical variance of the RTP data packet
interarrival time, measured in timestamp units and expressed as an
unsigned integer. The interarrival jitter J is defined to be the
mean deviation (smoothed absolute value) of the difference D in
packet spacing at the receiver compared to the sender for a pair
of packets. As shown in the equation below, this is equivalent to
the difference in the "relative transit time" for the two packets;
the relative transit time is the difference between a packet's RTP timestamp and the receiver's clock at the time of arrival, measured in the same units.
If Si is the RTP timestamp from packet i, and Ri is the time of arrival in RTP timestamp units for packet i, then for two packets i and j, D may be expressed as
D(i,j) = (Rj - Ri) - (Sj - Si) = (Rj - Sj) - (Ri - Si)
The interarrival jitter SHOULD be calculated continuously as each data packet i is received from source SSRC_n, using this difference D for that packet and the previous packet i-1 in order of arrival (not necessarily in sequence), according to the formula
J(i) = J(i-1) + (|D(i-1,i)| - J(i-1))/16
Whenever a reception report is issued, the current value of J is sampled.
The jitter calculation MUST conform to the formula specified here in order to allow profile-independent monitors to make valid interpretations of reports coming from different implementations. This algorithm is the optimal first-order estimator and the gain parameter 1/16 gives a good noise reduction ratio while maintaining a reasonable rate of convergence [22]. A sample implementation is shown in Appendix A.8. See Section 6.4.4 for a discussion of the effects of varying packet duration and delay before transmission.
last SR timestamp (LSR): 32 bits
The middle 32 bits out of 64 in the NTP timestamp (as explained in
Section 4) received as part of the most recent RTCP sender report
(SR) packet from source SSRC_n. If no SR has been received yet,
the field is set to zero.
delay since last SR (DLSR): 32 bits
The delay, expressed in units of 1/65536 seconds, between
receiving the last SR packet from source SSRC_n and sending this
reception report block. If no SR packet has been received yet
from SSRC_n, the DLSR field is set to zero.
Let SSRC_r denote the receiver issuing this receiver report. Source SSRC_n can compute the round-trip propagation delay to SSRC_r by recording the time A when this reception report block is received. It calculates the total round-trip time A-LSR using the last SR timestamp (LSR) field, and then subtracting this field to leave the round-trip propagation delay as (A - LSR - DLSR). This
is illustrated in Fig. 2. Times are shown in both a hexadecimal representation of the 32-bit fields and the equivalent floating- point decimal representation. Colons indicate a 32-bit field divided into a 16-bit integer part and 16-bit fraction part.
This may be used as an approximate measure of distance to cluster receivers, although some links have very asymmetric delays.
[10 Nov 1995 11:33:25.125 UTC] [10 Nov 1995 11:33:36.5 UTC]
n SR(n) A=b710:8000 (46864.500 s)
---------------------------------------------------------------->
v ^
ntp_sec =0xb44db705 v ^ dlsr=0x0005:4000 ( 5.250s)
ntp_frac=0x20000000 v ^ lsr =0xb705:2000 (46853.125s)
(3024992005.125 s) v ^
r v ^ RR(n)
---------------------------------------------------------------->
|<-DLSR->|
(5.250 s)
A 0xb710:8000 (46864.500 s)
DLSR -0x0005:4000 ( 5.250 s)
LSR -0xb705:2000 (46853.125 s)
-------------------------------
delay 0x0006:2000 ( 6.125 s)
Figure 2: Example for round-trip time computation
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of packet sender |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
1 | fraction lost | cumulative number of packets lost |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| extended highest sequence number received |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| interarrival jitter |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| last SR (LSR) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| delay since last SR (DLSR) |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
2 : ... :
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| profile-specific extensions |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The format of the receiver report (RR) packet is the same as that of the SR packet except that the packet type field contains the constant 201 and the five words of sender information are omitted (these are the NTP and RTP timestamps and sender's packet and octet counts). The remaining fields have the same meaning as for the SR packet.
An empty RR packet (RC = 0) MUST be put at the head of a compound RTCP packet when there is no data transmission or reception to report.
A profile SHOULD define profile-specific extensions to the sender report and receiver report if there is additional information that needs to be reported regularly about the sender or receivers. This method SHOULD be used in preference to defining another RTCP packet type because it requires less overhead:
The extension is a fourth section in the sender- or receiver-report packet which comes at the end after the reception report blocks, if any. If additional sender information is required, then for sender reports it would be included first in the extension section, but for receiver reports it would not be present. If information about receivers is to be included, that data SHOULD be structured as an array of blocks parallel to the existing array of reception report blocks; that is, the number of blocks would be indicated by the RC field.
It is expected that reception quality feedback will be useful not only for the sender but also for other receivers and third-party monitors. The sender may modify its transmissions based on the feedback; receivers can determine whether problems are local, regional or global; network managers may use profile-independent monitors that receive only the RTCP packets and not the corresponding RTP data packets to evaluate the performance of their networks for multicast distribution.
Cumulative counts are used in both the sender information and receiver report blocks so that differences may be calculated between any two reports to make measurements over both short and long time periods, and to provide resilience against the loss of a report. The difference between the last two reports received can be used to estimate the recent quality of the distribution. The NTP timestamp is included so that rates may be calculated from these differences over the interval between two reports. Since that timestamp is independent of the clock rate for the data encoding, it is possible to implement encoding- and profile-independent quality monitors.
An example calculation is the packet loss rate over the interval between two reception reports. The difference in the cumulative number of packets lost gives the number lost during that interval. The difference in the extended last sequence numbers received gives the number of packets expected during the interval. The ratio of these two is the packet loss fraction over the interval. This ratio should equal the fraction lost field if the two reports are consecutive, but otherwise it may not. The loss rate per second can be obtained by dividing the loss fraction by the difference in NTP timestamps, expressed in seconds. The number of packets received is the number of packets expected minus the number lost. The number of
packets expected may also be used to judge the statistical validity of any loss estimates. For example, 1 out of 5 packets lost has a lower significance than 200 out of 1000.
From the sender information, a third-party monitor can calculate the average payload data rate and the average packet rate over an interval without receiving the data. Taking the ratio of the two gives the average payload size. If it can be assumed that packet loss is independent of packet size, then the number of packets received by a particular receiver times the average payload size (or the corresponding packet size) gives the apparent throughput available to that receiver.
In addition to the cumulative counts which allow long-term packet loss measurements using differences between reports, the fraction lost field provides a short-term measurement from a single report. This becomes more important as the size of a session scales up enough that reception state information might not be kept for all receivers or the interval between reports becomes long enough that only one report might have been received from a particular receiver.
The interarrival jitter field provides a second short-term measure of network congestion. Packet loss tracks persistent congestion while the jitter measure tracks transient congestion. The jitter measure may indicate congestion before it leads to packet loss. The interarrival jitter field is only a snapshot of the jitter at the time of a report and is not intended to be taken quantitatively. Rather, it is intended for comparison across a number of reports from one receiver over time or from multiple receivers, e.g., within a single network, at the same time. To allow comparison across receivers, it is important the the jitter be calculated according to the same formula by all receivers.
Because the jitter calculation is based on the RTP timestamp which represents the instant when the first data in the packet was sampled, any variation in the delay between that sampling instant and the time the packet is transmitted will affect the resulting jitter that is calculated. Such a variation in delay would occur for audio packets of varying duration. It will also occur for video encodings because the timestamp is the same for all the packets of one frame but those packets are not all transmitted at the same time. The variation in delay until transmission does reduce the accuracy of the jitter calculation as a measure of the behavior of the network by itself, but it is appropriate to include considering that the receiver buffer must accommodate it. When the jitter calculation is used as a comparative measure, the (constant) component due to variation in delay until transmission subtracts out so that a change in the
network jitter component can then be observed unless it is relatively
small. If the change is small, then it is likely to be
inconsequential.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SDES items |
| ... |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
2 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SDES items |
| ... |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
The SDES packet is a three-level structure composed of a header and zero or more chunks, each of which is composed of items describing the source identified in that chunk. The items are described individually in subsequent sections.
version (V), padding (P), length:
As described for the SR packet (see Section 6.4.1).
packet type (PT): 8 bits
Contains the constant 202 to identify this as an RTCP SDES packet.
source count (SC): 5 bits
The number of SSRC/CSRC chunks contained in this SDES packet. A
value of zero is valid but useless.
Each chunk consists of an SSRC/CSRC identifier followed by a list of zero or more items, which carry information about the SSRC/CSRC. Each chunk starts on a 32-bit boundary. Each item consists of an 8- bit type field, an 8-bit octet count describing the length of the text (thus, not including this two-octet header), and the text itself. Note that the text can be no longer than 255 octets, but this is consistent with the need to limit RTCP bandwidth consumption.
The text is encoded according to the UTF-8 encoding specified in RFC 2279 [5]. US-ASCII is a subset of this encoding and requires no additional encoding. The presence of multi-octet encodings is indicated by setting the most significant bit of a character to a value of one.
Items are contiguous, i.e., items are not individually padded to a 32-bit boundary. Text is not null terminated because some multi- octet encodings include null octets. The list of items in each chunk MUST be terminated by one or more null octets, the first of which is interpreted as an item type of zero to denote the end of the list. No length octet follows the null item type octet, but additional null octets MUST be included if needed to pad until the next 32-bit boundary. Note that this padding is separate from that indicated by the P bit in the RTCP header. A chunk with zero items (four null octets) is valid but useless.
End systems send one SDES packet containing their own source identifier (the same as the SSRC in the fixed RTP header). A mixer sends one SDES packet containing a chunk for each contributing source from which it is receiving SDES information, or multiple complete SDES packets in the format above if there are more than 31 such sources (see Section 7).
The SDES items currently defined are described in the next sections. Only the CNAME item is mandatory. Some items shown here may be useful only for particular profiles, but the item types are all assigned from one common space to promote shared use and to simplify profile-independent applications. Additional items may be defined in a profile by registering the type numbers with IANA as described in Section 15.
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:
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 [6], RFC 1035 [7] and Section 2.1 of RFC 1123 [8]; or the standard ASCII representation of the host's numeric address on the interface used for the RTP communication. For example, the standard ASCII representation of an IP Version 4 address is "dotted decimal", also known as dotted quad, and for IP Version 6, addresses are textually represented as groups of hexadecimal digits separated by colons (with variations as detailed in RFC 3513 [23]). Other address types are expected to have ASCII representations that are mutually unique. The fully qualified domain name is more convenient for a human observer and may avoid the need to send a NAME item in addition, but it may be difficult or impossible to obtain reliably in some operating environments. Applications that may be run in such environments SHOULD use the ASCII representation of the address instead.
Examples are "doe@sleepy.example.com", "doe@192.0.2.89" or "doe@2201:056D::112E:144A:1E24" for a multi-user system. On a system with no user name, examples would be "sleepy.example.com", "192.0.2.89" or "2201:056D::112E:144A:1E24".
The user name SHOULD be in a form that a program such as "finger" or "