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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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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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-
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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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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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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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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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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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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
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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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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
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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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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
&