Network Working Group D. Hoffman
Request for Comments: 2038 G. Fernando
Category: Standards Track Sun Microsystems, Inc.
V. Goyal
Precept Software, Inc.
October 1996

RTP Payload Format for MPEG1/MPEG2 Video

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 memo describes a packetization scheme for MPEG video and audio
streams. The scheme proposed can be used to transport sUCh a video
or audio flow over the transport protocols supported by RTP. Two
approaches are described. The first is designed to support maximum
interoperability with MPEG System environments. The second is
designed to provide maximum compatibility with other RTP-encapsulated
media streams and future conference control work of the IETF.

1. Introduction

ISO/IEC JTC1/SC29 WG11 (also referred to as the MPEG committee) has
defined the MPEG1 standard (ISO/IEC 11172)[1] and the MPEG2 standard
(ISO/IEC 13818)[2]. This memo describes a packetization scheme to
transport MPEG video and audio streams using the Real-time Transport
Protocol (RTP), version 2 [3, 4].

The MPEG1 specification is defined in three parts: System, Video and
Audio. It is designed primarily for CD-ROM-based applications, and
is optimized for approximately 1.5 Mbits/sec combined data rates. The
video and audio portions of the specification describe the basic
format of the video or audio stream. These formats define the
Elementary Streams (ES). The MPEG1 System specification defines an
encapsulation of the ES that contains Presentation Time Stamps (PTS),
Decoding Time Stamps and System Clock references, and performs
multiplexing of MPEG1 compressed video and audio ES's with user data.

The MPEG2 specification is structured in a similar way. However, it
hasn't been restricted only to CD-ROM applications. The MPEG2 System
specification defines two system stream formats: the MPEG2 Transport
Stream (MTS) and the MPEG2 Program Stream (MPS). The MTS is tailored
for communicating or storing one or more programs of MPEG2 compressed
data and also other data in relatively error-prone environments. The
MPS is tailored for relatively error-free environments.

We seek to achieve interoperability among 4 types of end-systems in
the following specification. The 4 types are:

1. Transmitting Interworking Unit (TIU)

Receives MPEG information from a native MTS system for
distribution over packet networks using a native RTP-based
system layer (such as an IP-based internetwork). Examples:
real-time encoder, MTS satellite link to Internet, video
server with MTS-encoded source material.

2. Receiving Interworking Unit (RIU)

Receives MPEG information in real time from an RTP-based
network for forwarding to a native MTS environment.
Examples: Internet-based video server to MTS-based cable
distribution plant.

3. Transmitting Internet End-System (TAES)

Transmits MPEG information generated or stored within the
internet end-system itself, or received from internet-based
computer networks. Example: video server.

4. Receiving Internet End-System (RAES)

Receives MPEG information over an RTP-based internet for
consumption at the internet end-system or forwarding to
traditional computer network. Example: desktop PC or
workstation viewing training video.

Each of the 2 types of transmitters must work with each of the 2
types of receivers. Because it is probable that the TAES, and
certain that the RAES, will be based on existing and planned
internet-connected computers, it is highly desirable for the
interoperable protocol to be based on RTP.

Because of the range of applications that might employ MPEG streams,
we propose to define two payload formats.

Much interest in the MPEG community is in the use of one of the MPEG
System encodings, and hence, in Section 2 we propose encapsulations
of MPEG1 System streams and MPEG2 Transport and Program Streams with
RTP. This profile supports the full semantics of MPEG System and
offers basic interoperability among all four end-system types.

When operating only among internet-based end-systems (i.e., TAES and
RAES) a payload format that provides greater compatibility with the
Internet architecture is desired, deferring some of the system issues
to other protocols being defined in the Internet community (such as
the MMUSIC WG). In Section 3 we propose an encapsulation of
compressed video and audio data (referred to in MPEG documentation as
"Elementary Streams" (ES)) complying with either MPEG1 or MPEG2.
Here, neither of the System standards of MPEG1 or MPEG2 are utilized.
The ES's are directly encapsulated with RTP.

Throughout this specification, we make extensive use of MPEG
terminology. The reader should consult the primary MPEG references
for definitive descriptions of this terminology.

2. Encapsulation of MPEG System and Transport Streams

Each RTP packet will contain a timestamp derived from the sender's
90KHz clock reference. This clock is synchronized to the system
stream Program Clock Reference (PCR) or System Clock Reference (SCR)
and represents the target transmission time of the first byte of the
packet payload. The RTP timestamp will not be passed to the MPEG
decoder. This use of the timestamp is somewhat different than
normally is the case in RTP, in that it is not considered to be the
media display or presentation timestamp. The primary purposes of the
RTP timestamp will be to estimate and reduce any network-induced
jitter and to synchronize relative time drift between the transmitter
and receiver.

For MPEG2 Transport Streams the RTP payload will contain an integral
number of MPEG transport packets. To avoid end system
inefficiencies, data from multiple small MTS packets (normally fixed
in size at 188 bytes) are aggregated into a single RTP packet. The
number of transport packets contained is computed by dividing RTP
payload length by the length of an MTS packet (188).

For MPEG2 Program streams and MPEG1 system streams there are no
packetization restrictions; these streams are treated as a packetized
stream of bytes.

2.1 RTP header usage

The RTP header fields are used as follows:

Payload Type: Distinct payload types should be assigned for
of MPEG1 System Streams, MPEG2 Program Streams and MPEG2
Transport Streams. See [4] for payload type assignments.

M bit: Set to 1 whenever the timestamp is discontinuous
(such as might happen when a sender switches from one data
source to another). This allows the receiver and any
intervening RTP mixers or translators that are synchronizing
to the flow to ignore the difference between this timestamp
and any previous timestamp in their clock phase detectors.

timestamp: 32 bit 90K Hz timestamp representing the target
transmission time for the first byte of the packet.

3. Encapsulation of MPEG Elementary Streams

The following ES types may be encapsulated directly in RTP:

(a) MPEG1 Video (ISO/IEC 11172-2)
(b) MPEG2 Video (ISO/IEC 13818-2)
(c) MPEG1 Audio (ISO/IEC 11172-3)
(d) MPEG2 Audio (ISO/IEC 13818-3)

A distinct RTP payload type is assigned to MPEG1/MPEG2 Video and
MPEG1/MPEG2 Audio, respectively. Further indication as to whether the
data is MPEG1 or MPEG2 need not be provided in the RTP or MPEG-
specific headers of this encapsulation, as this information is
available in the ES headers.

Presentation Time Stamps (PTS) of 32 bits with an accuracy of 90 kHz
shall be carried in the fixed RTP header. All packets that make up a
audio or video frame shall have the same time stamp.

3.1 MPEG Video elementary streams

MPEG1 Video can be distinguished from MPEG2 Video at the video
sequence header, i.e. for MPEG2 Video a sequence_header() is followed
by sequence_extension(). The particular profile and level of MPEG2
Video (MAIN_Profile@MAIN_Level, HIGH_Profile@HIGH_Level, etc) are
determined by the profile_and_level_indicator field of the
sequence_extension header of MPEG2 Video.

The MPEG bit-stream semantics were designed for relatively error-free
environments, and there is significant amount of dependency (both

temporal and spatial) within the stream such that loss of some data
make other uncorrupted data useless. The format as defined in this
encapsulation uses application layer framing information plus
additional information in the RTP stream-specific header to allow for
certain recovery mechanisms. Appendix 1 suggests several recovery
strategies based on the properties of this encapsulation.

Since MPEG pictures can be large, they will normally be fragmented
into packets of size less than a typical LAN/WAN MTU. The following
fragmentation rules apply:

1. The MPEG Video_Sequence_Header, when present, will always
be at the beginning of an RTP payload.
2. An MPEG GOP_header, when present, will always be at the
beginning of the RTP payload, or will follow a
Video_Sequence_Header.
3. An MPEG Picture_Header, when present, will always be at the
beginning of a RTP payload, or will follow a GOP_header.

Each ES header must be completely contained within the packet.
Consequently, a minimum RTP payload size of 261 bytes must be
supported to contain the largest single header defined in the ES
(that is, the extension_data() header containing the
quant_matrix_extension()). Otherwise, there are no restrictions on
where headers may appear within packet payloads.

In MPEG, each picture is made up of one or more "slices," and a slice
is intended to be the unit of recovery from data loss or corruption.
An MPEG-compliant decoder will normally advance to the beginning of
next slice whenever an error is encountered in the stream. MPEG
slice begin and end bits are provided in the encapsulation header to
facilitate this.

The beginning of a slice must either be the first data in a packet
(after any MPEG ES headers) or must follow after some integral number
of slices in a packet. This requirement insures that the beginning
of the next slice after one with a missing packet can be found
without requiring that the receiver scan the packet contents. Slices
may be fragmented across packets as long as all the above rules are
met.

An implementation based on this encapsulation assumes that the
Video_Sequence_Header is repeated periodically in the MPEG bit-
stream. In practice (though not required by MPEG standard) this is
used to allow channel switching and to receive and start decoding a
continuously relayed MPEG bit-stream at arbitrary points in the media
stream. It is suggested that when playing back from an MPEG stream
from a file format (where the Video_Sequence_Header may only be

represented at the beginning of the stream) that the first
Video_Sequence_Header (preceded by an end-of-stream indicator) be
saved by the packetizer for periodic injection in to the network
stream.

3.2 MPEG Audio elementary streams

MPEG1 Audio can be distinguished from MPEG2 Audio from the MPEG
ancillary_data() header. For either MPEG1 or MPEG2 Audio, distinct
Presentation Time Stamps may be present for frames which correspond
to either 384 samples for Layer-I, or 1152 samples for Layer-II or
Layer-III. The actual number of bytes required to represent this
number of samples will vary depending on the encoder parameters.

Multiple audio frames may be encapsulated within one RTP packet. In
this case, an integral number of audio frames must be contained
within the packet and the fragmentation header defined in Section 3.5
shall be set to 0.

Also, if relatively short packets are to be used, one frame may be so
large that it may straddle multiple RTP packets. For example, for
Layer-II MPEG audio sampled at a rate of 44.1 KHz each frame would
represent a time slot of 26.1 msec. At this sampling rate if the
compressed bit-rate is 384 kbits/sec (i.e. 48 kBytes/sec) then the
average audio frame size would be 1.25 KBytes. If packets were to be
500 Bytes long, then each audio frame would straddle 3 RTP packets.
The audio fragmentation indicator header (See Section 3.5) shall be
present for an MPEG1/2 Audio payload type to provide for this
fragmentation.

3.3 RTP Fixed Header for MPEG ES encapsulation

The RTP header fields are used as follows:

Payload Type: Distinct payload types should be assigned
for video elementary streams and audio elementary streams.
See [4] for payload type assignments.

M bit: For video, set to 1 on packet containing MPEG frame
end code, 0 otherwise. For audio, set to 1 on first packet
of a "talk-spurt," 0 otherwise.

PT: MPEG video or audio stream ID.

timestamp: 32-bit 90K Hz timestamp representing presentation
time of MPEG picture or audio frame. Same for all packets
that make up a picture or audio frame. May not be
monotonically increasing in video stream if B pictures

present in stream. For packets that contain only a video
sequence and/or GOP header, the timestamp is that of the
subsequent picture.

3.4 MPEG Video-specific header

This header shall be attached to each RTP packet after the RTP fixed
header.

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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
MBZ TR MBZSBE P BFC FFC
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
FBV FFV

MBZ: Unused. Must be set to zero in current
specification. This space is reserved for future use.

TR: Temporal-Reference (10 bits). The temporal reference of
the current picture within the current GOP. This value
ranges from 0-1023 and is constant for all RTP packets of a
given picture.

MBZ: Unused. Must be set to zero in current
specification. This space is reserved for future use.

S: Sequence-header-present (1 bit). Normally 0 and set to 1 at
the occurrence of each MPEG sequence header. Used to
detect presence of sequence header in RTP packet.

B: Beginning-of-slice (BS) (1 bit). Set when the start of the
packet payload is a slice start code, or when a slice start
code is preceded only by one or more of a
Video_Sequence_Header, GOP_header and/or Picture_Header.

E: End-of-slice (ES) (1 bit). Set when the last byte of the
payload is the end of an MPEG slice.

P: Picture-Type (3 bits). I (1), P (2), B (3) or D (4). This
value is constant for each RTP packet of a given picture.
Value 000B is forbidden and 101B - 111B are reserved to
support future extensions to the MPEG ES specification.

FBV: full_pel_backward_vector
BFC: backward_f_code
FFV: full_pel_forward_vector
FFC: forward_f_code

OBTained from the most recent picture header, and are
constant for each RTP packet of a given picture. None of
these values are used for I frames and must be set to zero
in the RTP header. For P frames only the last two values
are present and FBV and BFC must be set to zero in the RTP
header. For B frames all the four values are present.

3.5 MPEG Audio-specific header

This header shall be attached to each RTP packet at the start of the
payload and after any RTP headers for an MPEG1/2 Audio payload type.

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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
MBZ Frag_offset
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Frag_offset: Byte offset into the audio frame for the data
in this packet.

Appendix 1. Error Recovery and Resynchronization Strategies.

The following error recovery and resynchronization strategies are
intended to be guidelines only. A compliant receiver is free to
employ alternative (or no) strategies.

When initially decoding an RTP-encapsulated MPEG Elementary Stream,
the receiver may discard all packets until the Sequence-header-
present bit is set to 1. At this point, sufficient state information
is contained in the stream to allow processing by an MPEG decoder.

Loss of packets containing the GOP_header and/or Picture_Header are
detected by an uneXPected change in the Temporal-Reference and
Picture-Type values. Consider the following example GOP sequence:

In display order: 0B 1B 2I 3B 4B 5P 6B 7B 8P GOP_HDR 0B ...
In stream order: 2I 0B 1B 5P 3B 4B 8P 6B 7B GOP_HDR 2I ...

Consider also two counters:

ref_pic_temp (Reference Picture (I,P) Temporal Reference)
dep_pic_temp (Dependent Picture (B) Temporal Reference)

At each GOP beginning, set these counters to the temporal reference
value of the corresponding picture type. For our example GOP
sequence, ref_pic_temp = 2 and dep_pic_temp = 0. Keep incrementing
BOTH counters by unity with each following picture. Ref_pic_temp
should match the temporal references of the I and P frames, and
dep_pic_temp should match the temporal references of the B frames.

dep_pic_temp: - 0 1 2 3 4 5 6 7 8 9
In stream order: 2I 0B 1B 5P 3B 4B 8P 6B 7B GOP_H 2I 0B 1B ...
ref_pic_temp: 2 3 4 5 6 7 8 9 10 ^ 11
-------------------------- ^
Match Drop
Mismatch
in ref_pic_temp

The loss of a GOP header can be detected by matching the appropriate
counter (based on picture type) to the temporal reference value. A
mismatch indicates a lost GOP header. If desired, a GOP header can be
re-constructed using a "null" time_code, repeating the closed_gop
flag from previous GOP headers, and setting the broken_link flag to
1.

The loss of a Picture_Header can also be detected by a mismatch in
the Temporal Reference contained in the RTP packet from the
appropriate dep_pic_temp or ref_pic_temp counters at the receiver.

After scanning to the next Beginning-of-slice the Picture_Header is
reconstructed from the P, TR, FBV, BFC, FFV and FFC contained in that
packet, and from stream-dependent default values.

Any time an RTP packet is lost (as indicated by a gap in the RTP
sequence number), the receiver may discard all packets until the
Beginning-of-slice bit is set. At this point, sufficient state
information is contained in the stream to allow processing by an MPEG
decoder starting at the next slice boundary (possibly after
reconstruction of the GOP_header and/or Picture_Header as described
above).

References

[1] ISO/IEC International Standard 11172; "Coding of moving pictures
and associated audio for digital storage media up to about 1,5
Mbits/s", November 1993.

[2] ISO/IEC International Standard 13818; "Generic coding of moving
pictures and associated audio information", November 1994.

[3] H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications",
RFC1889, January 1996.

[4] H. Schulzrinne, "RTP Profile for Audio and Video Conferences
with Minimal Control", RFC1890, January 1996.

Authors' Addresses

Gerard Fernando
Sun Microsystems, Inc.
Mail-stop UMPK14-305
2550 Garcia Avenue
Mountain View, California 94043-1100
USA

Phone: +1 415-786-6373
EMail: gerard.fernando@eng.sun.com

Vivek Goyal
Precept Software, Inc.
1072 Arastradero Rd,
Palo Alto, CA 94304
USA

Phone: +1 415-845-5200
EMail: goyal@precept.com

Don Hoffman
Sun Microsystems, Inc.
Mail-stop UMPK14-305
2550 Garcia Avenue
Mountain View, California 94043-1100
USA

Phone: +1 503-297-1580
EMail: don.hoffman@eng.sun.com