draft-ietf-avt-profile-new-12.txt   draft-ietf-avt-profile-new-13.txt 
Internet Engineering Task Force AVT WG Internet Engineering Task Force AVT WG
Internet Draft Schulzrinne/Casner Internet Draft Schulzrinne/Casner
draft-ietf-avt-profile-new-12.txt Columbia U./Packet Design draft-ietf-avt-profile-new-13.txt Columbia U./Packet Design
November 20, 2001 March 2, 2003
Expires: May 2002 Expires: September 2003
RTP Profile for Audio and Video Conferences with Minimal Control RTP Profile for Audio and Video Conferences with Minimal Control
STATUS OF THIS MEMO STATUS OF THIS MEMO
This document is an Internet-Draft and is in full conformance with This document is an Internet-Draft and is subject to all provisions
all provisions of Section 10 of RFC2026. of Section 10 of RFC 2026.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet- other groups may also distribute working documents as Internet-
Drafts. Drafts.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress". material or to cite them other than as "work in progress".
The list of current Internet-Drafts can be accessed at The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt http://www.ietf.org/1id-abstracts.html
To view the list Internet-Draft Shadow Directories, see The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html. http://www.ietf.org/shadow.html
Abstract Abstract
This memorandum is a revision of RFC 1890 in preparation for
advancement from Proposed Standard to Draft Standard status.
This document describes a profile called "RTP/AVP" for the use of the This document describes a profile called "RTP/AVP" for the use of the
real-time transport protocol (RTP), version 2, and the associated real-time transport protocol (RTP), version 2, and the associated
control protocol, RTCP, within audio and video multiparticipant control protocol, RTCP, within audio and video multiparticipant
conferences with minimal control. It provides interpretations of conferences with minimal control. It provides interpretations of
generic fields within the RTP specification suitable for audio and generic fields within the RTP specification suitable for audio and
video conferences. In particular, this document defines a set of video conferences. In particular, this document defines a set of
default mappings from payload type numbers to encodings. default mappings from payload type numbers to encodings.
This document also describes how audio and video data may be carried This document also describes how audio and video data may be carried
within RTP. It defines a set of standard encodings and their names within RTP. It defines a set of standard encodings and their names
when used within RTP. The descriptions provide pointers to reference when used within RTP. The descriptions provide pointers to reference
implementations and the detailed standards. This document is meant as implementations and the detailed standards. This document is meant
an aid for implementors of audio, video and other real-time as an aid for implementors of audio, video and other real-time
multimedia applications. multimedia applications.
This memorandum obsoletes RFC 1890. It is mostly backwards-
compatible except for functions removed because two interoperable
implementations were not found. The additions to RFC 1890 codify
existing practice in the use of payload formats under this profile
and include new payload formats defined since RFC 1890 was published.
Contents Contents
1 Introduction ........................................ 3 1 Introduction ........................................ 3
1.1 Terminology ......................................... 3 1.1 Terminology ......................................... 4
2 RTP and RTCP Packet Forms and Protocol Behavior ..... 4 2 RTP and RTCP Packet Forms and Protocol Behavior ..... 4
3 IANA Considerations ................................. 6 3 Registering Additional Encodings .................... 6
3.1 Registering Additional Encodings .................... 6
4 Audio ............................................... 8 4 Audio ............................................... 8
4.1 Encoding-Independent Rules .......................... 8 4.1 Encoding-Independent Rules .......................... 8
4.2 Operating Recommendations ........................... 9 4.2 Operating Recommendations ........................... 10
4.3 Guidelines for Sample-Based Audio Encodings ......... 10 4.3 Guidelines for Sample-Based Audio Encodings ......... 10
4.4 Guidelines for Frame-Based Audio Encodings .......... 10 4.4 Guidelines for Frame-Based Audio Encodings .......... 11
4.5 Audio Encodings ..................................... 11 4.5 Audio Encodings ..................................... 12
4.5.1 DVI4 ................................................ 12 4.5.1 DVI4 ................................................ 13
4.5.2 G722 ................................................ 13 4.5.2 G722 ................................................ 14
4.5.3 G723 ................................................ 13 4.5.3 G723 ................................................ 14
4.5.4 G726-40, G726-32, G726-24, and G726-16 .............. 17 4.5.4 G726-40, G726-32, G726-24, and G726-16 .............. 18
4.5.5 G728 ................................................ 18 4.5.5 G728 ................................................ 19
4.5.6 G729 ................................................ 19 4.5.6 G729 ................................................ 20
4.5.7 G729D and G729E ..................................... 21 4.5.7 G729D and G729E ..................................... 22
4.5.8 GSM ................................................. 24 4.5.8 GSM ................................................. 25
4.5.8.1 General Packaging Issues ............................ 24 4.5.9 GSM-EFR ............................................. 26
4.5.8.2 GSM variable names and numbers ...................... 25 4.5.10 L8 .................................................. 26
4.5.9 GSM-EFR ............................................. 25 4.5.11 L16 ................................................. 26
4.5.10 L8 .................................................. 25 4.5.12 LPC ................................................. 28
4.5.11 L16 ................................................. 25 4.5.13 MPA ................................................. 29
4.5.12 LPC ................................................. 27 4.5.14 PCMA and PCMU ....................................... 29
4.5.13 MPA ................................................. 28 4.5.15 QCELP ............................................... 29
4.5.14 PCMA and PCMU ....................................... 28 4.5.16 RED ................................................. 30
4.5.15 QCELP ............................................... 28 4.5.17 VDVI ................................................ 30
4.5.16 RED ................................................. 28 5 Video ............................................... 30
4.5.17 VDVI ................................................ 29 5.1 CelB ................................................ 31
5 Video ............................................... 29 5.2 JPEG ................................................ 31
5.1 CelB ................................................ 30 5.3 H261 ................................................ 31
5.2 JPEG ................................................ 30 5.4 H263 ................................................ 31
5.3 H261 ................................................ 30 5.5 H263-1998 ........................................... 32
5.4 H263 ................................................ 30 5.6 MPV ................................................. 32
5.5 H263-1998 ........................................... 31 5.7 MP2T ................................................ 32
5.6 MPV ................................................. 31 5.8 nv .................................................. 32
5.7 MP2T ................................................ 31 6 Payload Type Definitions ............................ 33
5.8 nv .................................................. 31 7 RTP over TCP and Similar Byte Stream Protocols ...... 33
6 Payload Type Definitions ............................ 31 8 Port Assignment ..................................... 35
7 RTP over TCP and Similar Byte Stream Protocols ...... 32 A Changes from RFC 1890 ............................... 36
8 Port Assignment ..................................... 32 B Security Considerations ............................. 38
9 Changes from RFC 1890 ............................... 34 C IANA Considerations ................................. 39
10 Security Considerations ............................. 37 D References .......................................... 39
11 Full Copyright Statement ............................ 37 E Current Locations of Related Resources .............. 41
12 Acknowledgments ..................................... 38 F Acknowledgments ..................................... 43
13 Addresses of Authors ................................ 38 G Addresses of Authors ................................ 43
H Intellectual Property Rights Statement .............. 43
I Full Copyright Statement ............................ 44
1 Introduction 1 Introduction
[Note to the RFC Editor: This paragraph and the first paragraph of [Note to the RFC Editor: This paragraph is to be deleted when this
the Abstract are to be deleted when this draft is published as an draft is published as an RFC. All RFC XXXX should be filled in with
RFC. All RFC XXXX should be filled in with the number of the RTP the number of the RTP specification RFC submitted for Draft Standard
specification RFC submitted for Draft Standard status, and all RFC status, and all RFC YYYY should be filled in with the number of the
YYYY should be filled in with the number of the draft specifying MIME draft specifying MIME registration of RTP payload types as it is
registration of RTP payload types as it is submitted for Proposed submitted for Proposed Standard status. These latter references are
Standard status. These latter references are intended to be non- intended to be non-normative as this Profile may be used
normative as this Profile may be used independently of the MIME independently of the MIME registrations.]
registrations.]
This profile defines aspects of RTP left unspecified in the RTP This profile defines aspects of RTP left unspecified in the RTP
Version 2 protocol definition (RFC XXXX) [2]. This profile is Version 2 protocol definition (RFC XXXX) [1]. This profile is
intended for the use within audio and video conferences with minimal intended for the use within audio and video conferences with minimal
session control. In particular, no support for the negotiation of session control. In particular, no support for the negotiation of
parameters or membership control is provided. The profile is expected parameters or membership control is provided. The profile is
to be useful in sessions where no negotiation or membership control expected to be useful in sessions where no negotiation or membership
are used (e.g., using the static payload types and the membership control are used (e.g., using the static payload types and the
indications provided by RTCP), but this profile may also be useful in membership indications provided by RTCP), but this profile may also
conjunction with a higher-level control protocol. be useful in conjunction with a higher-level control protocol.
Use of this profile may be implicit in the use of the appropriate Use of this profile may be implicit in the use of the appropriate
applications; there may be no explicit indication by port number, applications; there may be no explicit indication by port number,
protocol identifier or the like. Applications such as session protocol identifier or the like. Applications such as session
directories may use the name for this profile specified in Section 3. directories may use the name for this profile specified in
Appendix C.
Other profiles may make different choices for the items specified Other profiles may make different choices for the items specified
here. here.
This document also defines a set of encodings and payload formats for This document also defines a set of encodings and payload formats for
audio and video. These payload format descriptions are included here audio and video. These payload format descriptions are included here
only as a matter of convenience since they are too small to warrant only as a matter of convenience since they are too small to warrant
separate documents. Use of these payload formats is NOT REQUIRED to separate documents. Use of these payload formats is NOT REQUIRED to
use this profile. Only the binding of some of the payload formats to use this profile. Only the binding of some of the payload formats to
static payload type numbers in Tables 4 and 5 is normative. static payload type numbers in Tables 4 and 5 is normative.
skipping to change at page 4, line 4 skipping to change at page 4, line 8
here. here.
This document also defines a set of encodings and payload formats for This document also defines a set of encodings and payload formats for
audio and video. These payload format descriptions are included here audio and video. These payload format descriptions are included here
only as a matter of convenience since they are too small to warrant only as a matter of convenience since they are too small to warrant
separate documents. Use of these payload formats is NOT REQUIRED to separate documents. Use of these payload formats is NOT REQUIRED to
use this profile. Only the binding of some of the payload formats to use this profile. Only the binding of some of the payload formats to
static payload type numbers in Tables 4 and 5 is normative. static payload type numbers in Tables 4 and 5 is normative.
1.1 Terminology 1.1 Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [1] and document are to be interpreted as described in RFC 2119 [2] and
indicate requirement levels for implementations compliant with this indicate requirement levels for implementations compliant with this
RTP profile. RTP profile.
This draft defines the term media type as dividing encodings of audio This draft defines the term media type as dividing encodings of audio
and video content into three classes: audio, video and audio/video and video content into three classes: audio, video and audio/video
(interleaved). (interleaved).
2 RTP and RTCP Packet Forms and Protocol Behavior 2 RTP and RTCP Packet Forms and Protocol Behavior
The section "RTP Profiles and Payload Format Specification" of RFC The section "RTP Profiles and Payload Format Specification" of
XXXX enumerates a number of items that can be specified or modified RFC XXXX enumerates a number of items that can be specified or
in a profile. This section addresses these items. Generally, this modified in a profile. This section addresses these items.
profile follows the default and/or recommended aspects of the RTP Generally, this profile follows the default and/or recommended
specification. aspects of the RTP specification.
RTP data header: The standard format of the fixed RTP data RTP data header: The standard format of the fixed RTP data
header is used (one marker bit). header is used (one marker bit).
Payload types: Static payload types are defined in Section 6. Payload types: Static payload types are defined in Section 6.
RTP data header additions: No additional fixed fields are RTP data header additions: No additional fixed fields are
appended to the RTP data header. appended to the RTP data header.
RTP data header extensions: No RTP header extensions are RTP data header extensions: No RTP header extensions are
skipping to change at page 4, line 50 skipping to change at page 5, line 7
by this profile specification. by this profile specification.
RTCP report interval: The suggested constants are to be used for RTCP report interval: The suggested constants are to be used for
the RTCP report interval calculation. Sessions operating the RTCP report interval calculation. Sessions operating
under this profile MAY specify a separate parameter for the under this profile MAY specify a separate parameter for the
RTCP traffic bandwidth rather than using the default RTCP traffic bandwidth rather than using the default
fraction of the session bandwidth. The RTCP traffic fraction of the session bandwidth. The RTCP traffic
bandwidth MAY be divided into two separate session bandwidth MAY be divided into two separate session
parameters for those participants which are active data parameters for those participants which are active data
senders and those which are not. Following the senders and those which are not. Following the
recommendation in the RTP specification [2] that 1/4 of the recommendation in the RTP specification [1] that 1/4 of the
RTCP bandwidth be dedicated to data senders, the RTCP bandwidth be dedicated to data senders, the
RECOMMENDED default values for these two parameters would RECOMMENDED default values for these two parameters would
be 1.25% and 3.75%, respectively. For a particular session, be 1.25% and 3.75%, respectively. For a particular
the RTCP bandwidth for non-data-senders MAY be set to zero session, the RTCP bandwidth for non-data-senders MAY be set
when operating on unidirectional links or for sessions that to zero when operating on unidirectional links or for
don't require feedback on the quality of reception. The sessions that don't require feedback on the quality of
RTCP bandwidth for data senders SHOULD be kept non-zero so reception. The RTCP bandwidth for data senders SHOULD be
that sender reports can still be sent for inter-media kept non-zero so that sender reports can still be sent for
synchronization and to identify the source by CNAME. The inter-media synchronization and to identify the source by
means by which the one or two session parameters for RTCP CNAME. The means by which the one or two session
bandwidth are specified is beyond the scope of this memo. parameters for RTCP bandwidth are specified is beyond the
scope of this memo.
SR/RR extension: No extension section is defined for the RTCP SR SR/RR extension: No extension section is defined for the RTCP SR
or RR packet. or RR packet.
SDES use: Applications MAY use any of the SDES items described SDES use: Applications MAY use any of the SDES items described
in the RTP specification. While CNAME information MUST be in the RTP specification. While CNAME information MUST be
sent every reporting interval, other items SHOULD only be sent every reporting interval, other items SHOULD only be
sent every third reporting interval, with NAME sent seven sent every third reporting interval, with NAME sent seven
out of eight times within that slot and the remaining SDES out of eight times within that slot and the remaining SDES
items cyclically taking up the eighth slot, as defined in items cyclically taking up the eighth slot, as defined in
skipping to change at page 5, line 35 skipping to change at page 5, line 41
NAME is sent in RTCP packets 1, 4, 7, 10, 13, 16, 19, NAME is sent in RTCP packets 1, 4, 7, 10, 13, 16, 19,
while, say, EMAIL is used in RTCP packet 22. while, say, EMAIL is used in RTCP packet 22.
Security: The RTP default security services are also the default Security: The RTP default security services are also the default
under this profile. under this profile.
String-to-key mapping: No mapping is specified by this profile. String-to-key mapping: No mapping is specified by this profile.
Congestion: RTP and this profile may be used in the context of Congestion: RTP and this profile may be used in the context of
enhanced network service, for example, through Integrated enhanced network service, for example, through Integrated
Services (RFC 1633) [4] or Differentiated Services (RFC Services (RFC 1633) [4] or Differentiated Services
2475) [5], or they may be used with best effort service. (RFC 2475) [5], or they may be used with best effort
service.
If enhanced service is being used, RTP receivers SHOULD If enhanced service is being used, RTP receivers SHOULD
monitor packet loss to ensure that the service that was monitor packet loss to ensure that the service that was
requested is actually being delivered. If it is not, then requested is actually being delivered. If it is not, then
they SHOULD assume that they are receiving best-effort they SHOULD assume that they are receiving best-effort
service and behave accordingly. service and behave accordingly.
If best-effort service is being used, RTP receivers SHOULD If best-effort service is being used, RTP receivers SHOULD
monitor packet loss to ensure that the packet loss rate is monitor packet loss to ensure that the packet loss rate is
within acceptable parameters. Packet loss is considered within acceptable parameters. Packet loss is considered
skipping to change at page 6, line 31 skipping to change at page 6, line 38
preclude the use of these definitions when RTP is carried preclude the use of these definitions when RTP is carried
by other lower-layer protocols.) by other lower-layer protocols.)
Transport mapping: The standard mapping of RTP and RTCP to Transport mapping: The standard mapping of RTP and RTCP to
transport-level addresses is used. transport-level addresses is used.
Encapsulation: This profile leaves to applications the Encapsulation: This profile leaves to applications the
specification of RTP encapsulation in protocols other than specification of RTP encapsulation in protocols other than
UDP. UDP.
3 IANA Considerations 3 Registering Additional Encodings
The RTP specification establishes a registry of profile names for use
by higher-level control protocols, such as the Session Description
Protocol (SDP), RFC 2327 [6], to refer to transport methods. This
profile registers the name "RTP/AVP".
3.1 Registering Additional Encodings
This profile lists a set of encodings, each of which is comprised of This profile lists a set of encodings, each of which is comprised of
a particular media data compression or representation plus a payload a particular media data compression or representation plus a payload
format for encapsulation within RTP. Some of those payload formats format for encapsulation within RTP. Some of those payload formats
are specified here, while others are specified in separate RFCs. It are specified here, while others are specified in separate RFCs. It
is expected that additional encodings beyond the set listed here will is expected that additional encodings beyond the set listed here will
be created in the future and specified in additional payload format be created in the future and specified in additional payload format
RFCs. RFCs.
This profile also assigns to each encoding a short name which MAY be This profile also assigns to each encoding a short name which MAY be
skipping to change at page 7, line 15 skipping to change at page 7, line 15
In some contexts it may be useful to refer to these encodings in the In some contexts it may be useful to refer to these encodings in the
form of a MIME content-type. To facilitate this, RFC YYYY [7] form of a MIME content-type. To facilitate this, RFC YYYY [7]
provides registrations for all of the encodings names listed here as provides registrations for all of the encodings names listed here as
MIME subtype names under the "audio" and "video" MIME types through MIME subtype names under the "audio" and "video" MIME types through
the MIME registration procedure as specified in RFC 2048 [8]. the MIME registration procedure as specified in RFC 2048 [8].
Any additional encodings specified for use under this profile (or Any additional encodings specified for use under this profile (or
others) may also be assigned names registered as MIME subtypes with others) may also be assigned names registered as MIME subtypes with
the Internet Assigned Numbers Authority (IANA). This registry the Internet Assigned Numbers Authority (IANA). This registry
provides a means to insure that the names assigned to the additional provides a means to insure that the names assigned to the additional
encodings are kept unique. RFC YYYY specifies the information that is encodings are kept unique. RFC YYYY specifies the information that
required for the registration of RTP encodings. is required for the registration of RTP encodings.
In addition to assigning names to encodings, this profile also In addition to assigning names to encodings, this profile also
assigns static RTP payload type numbers to some of them. However, the assigns static RTP payload type numbers to some of them. However,
payload type number space is relatively small and cannot accommodate the payload type number space is relatively small and cannot
assignments for all existing and future encodings. During the early accommodate assignments for all existing and future encodings.
stages of RTP development, it was necessary to use statically During the early stages of RTP development, it was necessary to use
assigned payload types because no other mechanism had been specified statically assigned payload types because no other mechanism had been
to bind encodings to payload types. It was anticipated that non-RTP specified to bind encodings to payload types. It was anticipated
means beyond the scope of this memo (such as directory services or that non-RTP means beyond the scope of this memo (such as directory
invitation protocols) would be specified to establish a dynamic services or invitation protocols) would be specified to establish a
mapping between a payload type and an encoding. Now, mechanisms for dynamic mapping between a payload type and an encoding. Now,
defining dynamic payload type bindings have been specified in the mechanisms for defining dynamic payload type bindings have been
Session Description Protocol (SDP) and in other protocols such as specified in the Session Description Protocol (SDP) and in other
ITU-T recommendation H.323/H.245. These mechanisms associate the protocols such as ITU-T recommendation H.323/H.245. These mechanisms
registered name of the encoding/payload format, along with any associate the registered name of the encoding/payload format, along
additional required parameters such as the RTP timestamp clock rate with any additional required parameters such as the RTP timestamp
and number of channels, to a payload type number. This association clock rate and number of channels, to a payload type number. This
is effective only for the duration of the RTP session in which the association is effective only for the duration of the RTP session in
dynamic payload type binding is made. This association applies only which the dynamic payload type binding is made. This association
to the RTP session for which it is made, thus the numbers can be re- applies only to the RTP session for which it is made, thus the
used for different encodings in different sessions so the number numbers can be re-used for different encodings in different sessions
space limitation is avoided. so the number space limitation is avoided.
This profile reserves payload type numbers in the range 96-127 This profile reserves payload type numbers in the range 96-127
exclusively for dynamic assignment. Applications SHOULD first use exclusively for dynamic assignment. Applications SHOULD first use
values in this range for dynamic payload types. Those applications values in this range for dynamic payload types. Those applications
which need to define more than 32 dynamic payload types MAY bind which need to define more than 32 dynamic payload types MAY bind
codes below 96, in which case it is RECOMMENDED that unassigned codes below 96, in which case it is RECOMMENDED that unassigned
payload type numbers be used first. However, the statically assigned payload type numbers be used first. However, the statically assigned
payload types are default bindings and MAY be dynamically bound to payload types are default bindings and MAY be dynamically bound to
new encodings if needed. Redefining payload types below 96 may cause new encodings if needed. Redefining payload types below 96 may cause
incorrect operation if an attempt is made to join a session without incorrect operation if an attempt is made to join a session without
skipping to change at page 8, line 16 skipping to change at page 8, line 16
their own assignments of proprietary encodings to particular, fixed their own assignments of proprietary encodings to particular, fixed
payload types. payload types.
This specification establishes the policy that no additional static This specification establishes the policy that no additional static
payload types will be assigned beyond the ones defined in this payload types will be assigned beyond the ones defined in this
document. Establishing this policy avoids the problem of trying to document. Establishing this policy avoids the problem of trying to
create a set of criteria for accepting static assignments and create a set of criteria for accepting static assignments and
encourages the implementation and deployment of the dynamic payload encourages the implementation and deployment of the dynamic payload
type mechanisms. type mechanisms.
The final set of static payload type assignments is provided in
Tables 4 and 5. In particular, IANA should note that types 1 and 2
have been marked reserved and the set of "dyn" payload types included
has been updated. These changes are explained in Section 6 and
Appendix A.
4 Audio 4 Audio
4.1 Encoding-Independent Rules 4.1 Encoding-Independent Rules
Since the ability to suppress silence is one of the primary
motivations for using packets to transmit voice, the RTP header
carries both a sequence number and a timestamp to allow a receiver to
distinguish between lost packets and periods of time when no data was
transmitted. Discontiguous transmission (silence suppression) MAY be
used with any audio payload format. Receivers MUST assume that
senders may suppress silence unless this is restricted by signaling
specified elsewhere. (Even if the transmitter does not suppress
silence, the receiver should be prepared to handle periods when no
data is present since packets may be lost.)
Some payload formats (see Sections 4.5.3 and 4.5.6) define a "silence
insertion descriptor" or "comfort noise" frame to specify parameters
for artificial noise that may be generated during a period of silence
to approximate the background noise at the source. For other payload
formats, a generic Comfort Noise (CN) payload format is specified in
RFC 3389 [9]. When the CN payload format is used with another
payload format, different values in the RTP payload type field
distinguish comfort-noise packets from those of the selected payload
format.
For applications which send either no packets or occasional comfort- For applications which send either no packets or occasional comfort-
noise packets during silence, the first packet of a talkspurt, that noise packets during silence, the first packet of a talkspurt, that
is, the first packet after a silence period during which packets have is, the first packet after a silence period during which packets have
not been transmitted contiguously, SHOULD be distinguished by setting not been transmitted contiguously, SHOULD be distinguished by setting
the marker bit in the RTP data header to one. The marker bits in all the marker bit in the RTP data header to one. The marker bit in all
other packets is zero. The beginning of a talkspurt MAY be used to other packets is zero. The beginning of a talkspurt MAY be used to
adjust the playout delay to reflect changing network delays. adjust the playout delay to reflect changing network delays.
Applications without silence suppression MUST set the marker bit to Applications without silence suppression MUST set the marker bit to
zero. zero.
The RTP clock rate used for generating the RTP timestamp is The RTP clock rate used for generating the RTP timestamp is
independent of the number of channels and the encoding; it usually independent of the number of channels and the encoding; it usually
equals the number of sampling periods per second. For N-channel equals the number of sampling periods per second. For N-channel
encodings, each sampling period (say, 1/8000 of a second) generates N encodings, each sampling period (say, 1/8000 of a second) generates N
samples. (This terminology is standard, but somewhat confusing, as samples. (This terminology is standard, but somewhat confusing, as
skipping to change at page 9, line 8 skipping to change at page 9, line 34
l left l left
r right r right
c center c center
S surround S surround
F front F front
R rear R rear
channels description channel channels description channel
1 2 3 4 5 6 1 2 3 4 5 6
__________________________________________________ _________________________________________________
2 stereo l r 2 stereo l r
3 l r c 3 l r c
4 quadrophonic Fl Fr Rl Rr
4 l c r S 4 l c r S
5 Fl Fr Fc Sl Sr 5 Fl Fr Fc Sl Sr
6 l lc c r rc S 6 l lc c r rc S
Note: RFC 1890 defined two conventions for the ordering of
four audio channels. Since the ordering is indicated
implicitly by the number of channels, this was ambiguous.
In this revision, the order described as "quadrophonic" has
been eliminated to remove the ambiguity. This choice was
based on the observation that quadrophonic consumer audio
format did not become popular whereas surround-sound
subsequently has.
Samples for all channels belonging to a single sampling instant MUST Samples for all channels belonging to a single sampling instant MUST
be within the same packet. The interleaving of samples from different be within the same packet. The interleaving of samples from
channels depends on the encoding. General guidelines are given in different channels depends on the encoding. General guidelines are
Section 4.3 and 4.4. given in Section 4.3 and 4.4.
The sampling frequency SHOULD be drawn from the set: 8000, 11025, The sampling frequency SHOULD be drawn from the set: 8000, 11025,
16000, 22050, 24000, 32000, 44100 and 48000 Hz. (Older Apple 16000, 22050, 24000, 32000, 44100 and 48000 Hz. (Older Apple
Macintosh computers had a native sample rate of 22254.54 Hz, which Macintosh computers had a native sample rate of 22254.54 Hz, which
can be converted to 22050 with acceptable quality by dropping 4 can be converted to 22050 with acceptable quality by dropping 4
samples in a 20 ms frame.) However, most audio encodings are defined samples in a 20 ms frame.) However, most audio encodings are defined
for a more restricted set of sampling frequencies. Receivers SHOULD for a more restricted set of sampling frequencies. Receivers SHOULD
be prepared to accept multi-channel audio, but MAY choose to only be prepared to accept multi-channel audio, but MAY choose to only
play a single channel. play a single channel.
skipping to change at page 9, line 46 skipping to change at page 10, line 36
without additional negotiation. These guidelines are not intended to without additional negotiation. These guidelines are not intended to
restrict operating parameters for applications that can negotiate a restrict operating parameters for applications that can negotiate a
set of interoperable parameters, e.g., through a conference control set of interoperable parameters, e.g., through a conference control
protocol. protocol.
For packetized audio, the default packetization interval SHOULD have For packetized audio, the default packetization interval SHOULD have
a duration of 20 ms or one frame, whichever is longer, unless a duration of 20 ms or one frame, whichever is longer, unless
otherwise noted in Table 1 (column "ms/packet"). The packetization otherwise noted in Table 1 (column "ms/packet"). The packetization
interval determines the minimum end-to-end delay; longer packets interval determines the minimum end-to-end delay; longer packets
introduce less header overhead but higher delay and make packet loss introduce less header overhead but higher delay and make packet loss
more noticeable. For non-interactive applications such as lectures or more noticeable. For non-interactive applications such as lectures
for links with severe bandwidth constraints, a higher packetization or for links with severe bandwidth constraints, a higher
delay MAY be used. A receiver SHOULD accept packets representing packetization delay MAY be used. A receiver SHOULD accept packets
between 0 and 200 ms of audio data. (For framed audio encodings, a representing between 0 and 200 ms of audio data. (For framed audio
receiver SHOULD accept packets with a number of frames equal to 200 encodings, a receiver SHOULD accept packets with a number of frames
ms divided by the frame duration, rounded up.) This restriction equal to 200 ms divided by the frame duration, rounded up.) This
allows reasonable buffer sizing for the receiver. restriction allows reasonable buffer sizing for the receiver.
4.3 Guidelines for Sample-Based Audio Encodings 4.3 Guidelines for Sample-Based Audio Encodings
In sample-based encodings, each audio sample is represented by a In sample-based encodings, each audio sample is represented by a
fixed number of bits. Within the compressed audio data, codes for fixed number of bits. Within the compressed audio data, codes for
individual samples may span octet boundaries. An RTP audio packet may individual samples may span octet boundaries. An RTP audio packet
contain any number of audio samples, subject to the constraint that may contain any number of audio samples, subject to the constraint
the number of bits per sample times the number of samples per packet that the number of bits per sample times the number of samples per
yields an integral octet count. Fractional encodings produce less packet yields an integral octet count. Fractional encodings produce
than one octet per sample. less than one octet per sample.
The duration of an audio packet is determined by the number of The duration of an audio packet is determined by the number of
samples in the packet. samples in the packet.
For sample-based encodings producing one or more octets per sample, For sample-based encodings producing one or more octets per sample,
samples from different channels sampled at the same sampling instant samples from different channels sampled at the same sampling instant
SHOULD be packed in consecutive octets. For example, for a two- SHOULD be packed in consecutive octets. For example, for a two-
channel encoding, the octet sequence is (left channel, first sample), channel encoding, the octet sequence is (left channel, first sample),
(right channel, first sample), (left channel, second sample), (right (right channel, first sample), (left channel, second sample), (right
channel, second sample), .... For multi-octet encodings, octets channel, second sample), .... For multi-octet encodings, octets
skipping to change at page 10, line 40 skipping to change at page 11, line 28
The packing of sample-based encodings producing less than one octet The packing of sample-based encodings producing less than one octet
per sample is encoding-specific. per sample is encoding-specific.
The RTP timestamp reflects the instant at which the first sample in The RTP timestamp reflects the instant at which the first sample in
the packet was sampled, that is, the oldest information in the the packet was sampled, that is, the oldest information in the
packet. packet.
4.4 Guidelines for Frame-Based Audio Encodings 4.4 Guidelines for Frame-Based Audio Encodings
Frame-based encodings encode a fixed-length block of audio into Frame-based encodings encode a fixed-length block of audio into
another block of compressed data, typically also of fixed length. For another block of compressed data, typically also of fixed length.
frame-based encodings, the sender MAY choose to combine several such For frame-based encodings, the sender MAY choose to combine several
frames into a single RTP packet. The receiver can tell the number of such frames into a single RTP packet. The receiver can tell the
frames contained in an RTP packet, if all the frames have the same number of frames contained in an RTP packet, if all the frames have
length, by dividing the RTP payload length by the audio frame size the same length, by dividing the RTP payload length by the audio
which is defined as part of the encoding. This does not work when frame size which is defined as part of the encoding. This does not
carrying frames of different sizes unless the frame sizes are work when carrying frames of different sizes unless the frame sizes
relatively prime. If not, the frames MUST indicate their size. are relatively prime. If not, the frames MUST indicate their size.
For frame-based codecs, the channel order is defined for the whole For frame-based codecs, the channel order is defined for the whole
block. That is, for two-channel audio, right and left samples SHOULD block. That is, for two-channel audio, right and left samples SHOULD
be coded independently, with the encoded frame for the left channel be coded independently, with the encoded frame for the left channel
preceding that for the right channel. preceding that for the right channel.
All frame-oriented audio codecs SHOULD be able to encode and decode All frame-oriented audio codecs SHOULD be able to encode and decode
several consecutive frames within a single packet. Since the frame several consecutive frames within a single packet. Since the frame
size for the frame-oriented codecs is given, there is no need to use size for the frame-oriented codecs is given, there is no need to use
a separate designation for the same encoding, but with different a separate designation for the same encoding, but with different
skipping to change at page 12, line 18 skipping to change at page 13, line 7
"var." in the sampling rate column of Table 1) may be used with "var." in the sampling rate column of Table 1) may be used with
different sampling rates, resulting in different coded bit rates. different sampling rates, resulting in different coded bit rates.
When used with a sampling rate other than that for which a static When used with a sampling rate other than that for which a static
payload type is defined, non-RTP means beyond the scope of this memo payload type is defined, non-RTP means beyond the scope of this memo
MUST be used to define a dynamic payload type and MUST indicate the MUST be used to define a dynamic payload type and MUST indicate the
selected RTP timestamp clock rate, which is usually the same as the selected RTP timestamp clock rate, which is usually the same as the
sampling rate for audio. sampling rate for audio.
4.5.1 DVI4 4.5.1 DVI4
DVI4 is specified, with pseudo-code, in [9] as the IMA ADPCM wave DVI4 is specified, with pseudo-code, in [10] as the IMA ADPCM wave
type. type.
However, the encoding defined here as DVI4 differs in three respects However, the encoding defined here as DVI4 differs in three respects
from this recommendation: from this recommendation:
o The RTP DVI4 header contains the predicted value rather than o The RTP DVI4 header contains the predicted value rather than
the first sample value contained the IMA ADPCM block header. the first sample value contained the IMA ADPCM block header.
o IMA ADPCM blocks contain an odd number of samples, since the o IMA ADPCM blocks contain an odd number of samples, since the
first sample of a block is contained just in the header first sample of a block is contained just in the header
skipping to change at page 13, line 39 skipping to change at page 14, line 25
Even though the actual sampling rate for G.722 audio is 16000 Hz, the Even though the actual sampling rate for G.722 audio is 16000 Hz, the
RTP clock rate for the G722 payload format is 8000 Hz because that RTP clock rate for the G722 payload format is 8000 Hz because that
value was erroneously assigned in RFC 1890 and must remain unchanged value was erroneously assigned in RFC 1890 and must remain unchanged
for backward compatibility. The octet rate or sample-pair rate is for backward compatibility. The octet rate or sample-pair rate is
8000 Hz. 8000 Hz.
4.5.3 G723 4.5.3 G723
G723 is specified in ITU Recommendation G.723.1, "Dual-rate speech G723 is specified in ITU Recommendation G.723.1, "Dual-rate speech
coder for multimedia communications transmitting at 5.3 and 6.3 coder for multimedia communications transmitting at 5.3 and 6.3
kbit/s". The G.723.1 5.3/6.3 kbit/s codec was defined by the ITU-T as kbit/s". The G.723.1 5.3/6.3 kbit/s codec was defined by the ITU-T
a mandatory codec for ITU-T H.324 GSTN videophone terminal as a mandatory codec for ITU-T H.324 GSTN videophone terminal
applications. The algorithm has a floating point specification in applications. The algorithm has a floating point specification in
Annex B to G.723.1, a silence compression algorithm in Annex A to Annex B to G.723.1, a silence compression algorithm in Annex A to
G.723.1 and an encoded signal bit-error sensitivity specification in G.723.1 and an encoded signal bit-error sensitivity specification in
G.723.1 Annex C. G.723.1 Annex C.
This Recommendation specifies a coded representation that can be used This Recommendation specifies a coded representation that can be used
for compressing the speech signal component of multi-media services for compressing the speech signal component of multi-media services
at a very low bit rate. Audio is encoded in 30 ms frames, with an at a very low bit rate. Audio is encoded in 30 ms frames, with an
additional delay of 7.5 ms due to look-ahead. A G.723.1 frame can be additional delay of 7.5 ms due to look-ahead. A G.723.1 frame can be
one of three sizes: 24 octets (6.3 kb/s frame), 20 octets (5.3 kb/s one of three sizes: 24 octets (6.3 kb/s frame), 20 octets (5.3 kb/s
frame), or 4 octets. These 4-octet frames are called SID frames frame), or 4 octets. These 4-octet frames are called SID frames
(Silence Insertion Descriptor) and are used to specify comfort noise (Silence Insertion Descriptor) and are used to specify comfort noise
parameters. There is no restriction on how 4, 20, and 24 octet frames parameters. There is no restriction on how 4, 20, and 24 octet
are intermixed. The least significant two bits of the first octet in frames are intermixed. The least significant two bits of the first
the frame determine the frame size and codec type: octet in the frame determine the frame size and codec type:
bits content octets/frame bits content octets/frame
00 high-rate speech (6.3 kb/s) 24 00 high-rate speech (6.3 kb/s) 24
01 low-rate speech (5.3 kb/s) 20 01 low-rate speech (5.3 kb/s) 20
10 SID frame 4 10 SID frame 4
11 reserved 11 reserved
It is possible to switch between the two rates at any 30 ms frame It is possible to switch between the two rates at any 30 ms frame
boundary. Both (5.3 kb/s and 6.3 kb/s) rates are a mandatory part of boundary. Both (5.3 kb/s and 6.3 kb/s) rates are a mandatory part of
the encoder and decoder. The MIME registration for G723 in RFC YYYY the encoder and decoder. Receivers MUST accept both data rates and
[7] specifies parameters that MAY be used with MIME or SDP to MUST accept SID frames unless restriction of these capabilities has
restrict to a single data rate or to restrict the use of SID frames. been signaled. The MIME registration for G723 in RFC YYYY [7]
This coder was optimized to represent speech with near-toll quality specifies parameters that MAY be used with MIME or SDP to restrict to
at the above rates using a limited amount of complexity. a single data rate or to restrict the use of SID frames. This coder
was optimized to represent speech with near-toll quality at the above
rates using a limited amount of complexity.
The packing of the encoded bit stream into octets and the The packing of the encoded bit stream into octets and the
transmission order of the octets is specified in Rec. G.723.1 and is transmission order of the octets is specified in Rec. G.723.1 and is
the same as that produced by the G.723 C code reference the same as that produced by the G.723 C code reference
implementation. For the 6.3 kb/s data rate, this packing is implementation. For the 6.3 kb/s data rate, this packing is
illustrated as follows, where the header (HDR) bits are always "0 0" illustrated as follows, where the header (HDR) bits are always "0 0"
as shown in Fig. 1 to indicate operation at 6.3 kb/s, and the Z bit as shown in Fig. 1 to indicate operation at 6.3 kb/s, and the Z bit
is always set to zero. The diagrams show the bit packing in "network is always set to zero. The diagrams show the bit packing in "network
byte order," also known as big-endian order. The bits of each 32-bit byte order," also known as big-endian order. The bits of each 32-bit
word are numbered 0 to 31, with the most significant bit on the left word are numbered 0 to 31, with the most significant bit on the left
skipping to change at page 16, line 37 skipping to change at page 17, line 37
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| POS3 | POS2 | POS3 | PSIG1 | PSIG0 | PSIG3 | PSIG2 | | POS3 | POS2 | POS3 | PSIG1 | PSIG0 | PSIG3 | PSIG2 |
| | | | | | | | | | | | | | | |
|0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0|0 0 0 0|0 0 0 0|0 0 0 0| |0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0|0 0 0 0|0 0 0 0|0 0 0 0|
|3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|3 2 1 0|3 2 1 0|3 2 1 0|3 2 1 0| |3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|3 2 1 0|3 2 1 0|3 2 1 0|3 2 1 0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: G.723 (5.3 kb/s) bit packing Figure 2: G.723 (5.3 kb/s) bit packing
The packing of G.723.1 SID (silence) frames, which are indicated by The packing of G.723.1 SID (silence) frames, which are indicated by
the header (HDR) bits having the pattern "1 0", is depicted in Fig. the header (HDR) bits having the pattern "1 0", is depicted in
3. Fig. 3.
0 1 2 3 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 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| LPC |HDR| LPC | LPC | GAIN |LPC| | LPC |HDR| LPC | LPC | GAIN |LPC|
| | | | | | | | | | | | | |
|0 0 0 0 0 0|1 0|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2| |0 0 0 0 0 0|1 0|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2|
|5 4 3 2 1 0| |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2| |5 4 3 2 1 0| |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 3: G.723 SID mode bit packing Figure 3: G.723 SID mode bit packing
4.5.4 G726-40, G726-32, G726-24, and G726-16 4.5.4 G726-40, G726-32, G726-24, and G726-16
ITU-T Recommendation G.726 describes, among others, the algorithm ITU-T Recommendation G.726 describes, among others, the algorithm
recommended for conversion of a single 64 kbit/s A-law or mu-law PCM recommended for conversion of a single 64 kbit/s A-law or mu-law PCM
channel encoded at 8000 samples/sec to and from a 40, 32, 24, or 16 channel encoded at 8000 samples/sec to and from a 40, 32, 24, or 16
kbit/s channel. The conversion is applied to the PCM stream using an kbit/s channel. The conversion is applied to the PCM stream using an
Adaptive Differential Pulse Code Modulation (ADPCM) transcoding Adaptive Differential Pulse Code Modulation (ADPCM) transcoding
technique. The ADPCM representation consists of a series of codewords technique. The ADPCM representation consists of a series of
with a one-to-one correspondence to the samples in the PCM stream. codewords with a one-to-one correspondence to the samples in the PCM
The G726 data rates of 40, 32, 24, and 16 kbit/s have codewords of 5, stream. The G726 data rates of 40, 32, 24, and 16 kbit/s have
4, 3, and 2 bits respectively. codewords of 5, 4, 3, and 2 bits respectively.
The 16 and 24 kbit/s encodings do not provide toll quality speech. The 16 and 24 kbit/s encodings do not provide toll quality speech.
They are designed for used in overloaded Digital Circuit They are designed for used in overloaded Digital Circuit
Multiplication Equipment (DCME). ITU-T G.726 recommends that the 16 Multiplication Equipment (DCME). ITU-T G.726 recommends that the 16
and 24 kbit/s encodings should be alternated with higher data rate and 24 kbit/s encodings should be alternated with higher data rate
encodings to provide an average sample size of between 3.5 and 3.7 encodings to provide an average sample size of between 3.5 and 3.7
bits per sample. bits per sample.
The encodings of G.726 are here denoted as G726-40, G726-32, G726-24, The encodings of G.726 are here denoted as G726-40, G726-32, G726-24,
and G726-16. Prior to 1990, G721 described the 32 kbit/s ADPCM and G726-16. Prior to 1990, G721 described the 32 kbit/s ADPCM
encoding, and G723 described the 40, 32, and 16 kbit/s encodings. encoding, and G723 described the 40, 32, and 16 kbit/s encodings.
Thus, G726-32 designates the same algorithm as G721 in RFC 1890. Thus, G726-32 designates the same algorithm as G721 in RFC 1890.
A stream of G726 codewords contains no information on the encoding A stream of G726 codewords contains no information on the encoding
being used, therefore transitions between G726 encoding types is not being used, therefore transitions between G726 encoding types is not
permitted within a sequence of packed codewords. Applications MUST permitted within a sequence of packed codewords. Applications MUST
determine the encoding type of packed codewords from the RTP payload determine the encoding type of packed codewords from the RTP payload
identifier. identifier.
No payload-specific header information SHALL be included as part of No payload-specific header information SHALL be included as part of
the audio data. A stream of G726 codewords MUST be packed into octets the audio data. A stream of G726 codewords MUST be packed into
as follows: the first codeword is placed into the first octet such octets as follows: the first codeword is placed into the first octet
that the least significant bit of the codeword aligns with the least such that the least significant bit of the codeword aligns with the
significant bit in the octet, the second codeword is then packed so least significant bit in the octet, the second codeword is then
that its least significant bit coincides with the least significant packed so that its least significant bit coincides with the least
unoccupied bit in the octet. When a complete codeword cannot be significant unoccupied bit in the octet. When a complete codeword
placed into an octet, the bits overlapping the octet boundary are cannot be placed into an octet, the bits overlapping the octet
placed into the least significant bits of the next octet. Packing boundary are placed into the least significant bits of the next
MUST end with a completely packed final octet. The number of octet. Packing MUST end with a completely packed final octet. The
codewords packed will therefore be a multiple of 8, 2, 8, and 4 for number of codewords packed will therefore be a multiple of 8, 2, 8,
G726-40, G726-32, G726-24, and G726-16 respectively. An example of and 4 for G726-40, G726-32, G726-24, and G726-16 respectively. An
the packing scheme for G726-32 codewords is as shown: example of the packing scheme for G726-32 codewords is as shown,
where bit 7 is the least significant bit of the first octet, and bit
A3 is the least significant bit of the first codeword:
0 1 0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
|B B B B|A A A A|D D D D|C C C C| ... |B B B B|A A A A|D D D D|C C C C| ...
|0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3| |0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
An example of the packing scheme for G726-24 codewords is: An example of the packing scheme for G726-24 codewords follows, where
again bit 7 is the least significant bit of the first octet, and bit
A2 is the least significant bit of the first codeword:
0 1 2 0 1 2
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
|C C|B B B|A A A|F|E E E|D D D|C|H H H|G G G|F F| ... |C C|B B B|A A A|F|E E E|D D D|C|H H H|G G G|F F| ...
|1 2|0 1 2|0 1 2|2|0 1 2|0 1 2|0|0 1 2|0 1 2|0 1| |1 2|0 1 2|0 1 2|2|0 1 2|0 1 2|0|0 1 2|0 1 2|0 1|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
Note that the "little-endian" direction in which samples are packed
into octets in the G726-16, -24, -32 and -48 payload formats
specified here is consistent with ITU-T Recommendation X.420, but is
the opposite of what is specified in ITU-T Recommendation I.366.2
Annex E for ATM AAL2 transport. A second set of RTP payload formats
matching the packetization of I.366.2 Annex E and identified by MIME
subtypes AAL2-G726-16, -24, -32 and -48 will be specified in a
separate document.
4.5.5 G728 4.5.5 G728
G728 is specified in ITU-T Recommendation G.728, "Coding of speech at G728 is specified in ITU-T Recommendation G.728, "Coding of speech at
16 kbit/s using low-delay code excited linear prediction". 16 kbit/s using low-delay code excited linear prediction".
A G.278 encoder translates 5 consecutive audio samples into a 10-bit A G.278 encoder translates 5 consecutive audio samples into a 10-bit
codebook index, resulting in a bit rate of 16 kb/s for audio sampled codebook index, resulting in a bit rate of 16 kb/s for audio sampled
at 8,000 samples per second. The group of five consecutive samples is at 8,000 samples per second. The group of five consecutive samples
called a vector. Four consecutive vectors, labeled V1 to V4 (where V1 is called a vector. Four consecutive vectors, labeled V1 to V4
is to be played first by the receiver), build one G.728 frame. The (where V1 is to be played first by the receiver), build one G.728
four vectors of 40 bits are packed into 5 octets, labeled B1 through frame. The four vectors of 40 bits are packed into 5 octets, labeled
B5. B1 SHALL be placed first in the RTP packet. B1 through B5. B1 SHALL be placed first in the RTP packet.
Referring to the figure below, the principle for bit order is Referring to the figure below, the principle for bit order is
"maintenance of bit significance". Bits from an older vector are more "maintenance of bit significance". Bits from an older vector are
significant than bits from newer vectors. The MSB of the frame goes more significant than bits from newer vectors. The MSB of the frame
to the MSB of B1 and the LSB of the frame goes to LSB of B5. goes to the MSB of B1 and the LSB of the frame goes to LSB of B5.
1 2 3 3 1 2 3 3
0 0 0 0 9 0 0 0 0 9
++++++++++++++++++++++++++++++++++++++++ ++++++++++++++++++++++++++++++++++++++++
<---V1---><---V2---><---V3---><---V4---> vectors <---V1---><---V2---><---V3---><---V4---> vectors
<--B1--><--B2--><--B3--><--B4--><--B5--> octets <--B1--><--B2--><--B3--><--B4--><--B5--> octets
<------------- frame 1 ----------------> <------------- frame 1 ---------------->
In particular, B1 contains the eight most significant bits of V1, In particular, B1 contains the eight most significant bits of V1,
with the MSB of V1 being the MSB of B1. B2 contains the two least with the MSB of V1 being the MSB of B1. B2 contains the two least
skipping to change at page 19, line 35 skipping to change at page 20, line 34
the RTP packet and B5 last. the RTP packet and B5 last.
4.5.6 G729 4.5.6 G729
G729 is specified in ITU-T Recommendation G.729, "Coding of speech at G729 is specified in ITU-T Recommendation G.729, "Coding of speech at
8 kbit/s using conjugate structure-algebraic code excited linear 8 kbit/s using conjugate structure-algebraic code excited linear
prediction (CS-ACELP)". A reduced-complexity version of the G.729 prediction (CS-ACELP)". A reduced-complexity version of the G.729
algorithm is specified in Annex A to Rec. G.729. The speech coding algorithm is specified in Annex A to Rec. G.729. The speech coding
algorithms in the main body of G.729 and in G.729 Annex A are fully algorithms in the main body of G.729 and in G.729 Annex A are fully
interoperable with each other, so there is no need to further interoperable with each other, so there is no need to further
distinguish between them. The G.729 and G.729 Annex A codecs were distinguish between them. An implementation that signals or accepts
optimized to represent speech with high quality, where G.729 Annex A use of G729 payload format may implement either G.729 or G.729A
trades some speech quality for an approximate 50% complexity unless restricted by additional signaling specified elsewhere related
reduction [10]. See the next Section (4.5.7) for other data rates specifically to the encoding rather than the payload format. The
added in later G.729 Annexes. For all data rates, the sampling G.729 and G.729 Annex A codecs were optimized to represent speech
frequency (and RTP timestamp clock rate) is 8000 Hz. with high quality, where G.729 Annex A trades some speech quality for
an approximate 50% complexity reduction [11]. See the next Section
(4.5.7) for other data rates added in later G.729 Annexes. For all
data rates, the sampling frequency (and RTP timestamp clock rate) is
8000 Hz.
A voice activity detector (VAD) and comfort noise generator (CNG) A voice activity detector (VAD) and comfort noise generator (CNG)
algorithm in Annex B of G.729 is RECOMMENDED for digital simultaneous algorithm in Annex B of G.729 is RECOMMENDED for digital simultaneous
voice and data applications and can be used in conjunction with G.729 voice and data applications and can be used in conjunction with G.729
or G.729 Annex A. A G.729 or G.729 Annex A frame contains 10 octets, or G.729 Annex A. A G.729 or G.729 Annex A frame contains 10 octets,
while the G.729 Annex B comfort noise frame occupies 2 octets. The while the G.729 Annex B comfort noise frame occupies 2 octets.
MIME registration for G729 in RFC YYYY [7] specifies a parameter that Receivers MUST accept comfort noise frames if restriction of their
MAY be used with MIME or SDP to restrict the use of comfort noise use has not been signaled. The MIME registration for G729 in
frames. RFC YYYY [7] specifies a parameter that MAY be used with MIME or SDP
to restrict the use of comfort noise frames.
A G729 RTP packet may consist of zero or more G.729 or G.729 Annex A A G729 RTP packet may consist of zero or more G.729 or G.729 Annex A
frames, followed by zero or one G.729 Annex B frames. The presence of frames, followed by zero or one G.729 Annex B frames. The presence
a comfort noise frame can be deduced from the length of the RTP of a comfort noise frame can be deduced from the length of the RTP
payload. The default packetization interval is 20 ms (two frames), payload. The default packetization interval is 20 ms (two frames),
but in some situations it may be desirable to send 10 ms packets. An but in some situations it may be desirable to send 10 ms packets. An
example would be a transition from speech to comfort noise in the example would be a transition from speech to comfort noise in the
first 10 ms of the packet. For some applications, a longer first 10 ms of the packet. For some applications, a longer
packetization interval may be required to reduce the packet rate. packetization interval may be required to reduce the packet rate.
The transmitted parameters of a G.729/G.729A 10-ms frame, consisting The transmitted parameters of a G.729/G.729A 10-ms frame, consisting
of 80 bits, are defined in Recommendation G.729, Table 8/G.729. The of 80 bits, are defined in Recommendation G.729, Table 8/G.729. The
mapping of the these parameters is given below in Fig. 4. The mapping of the these parameters is given below in Fig. 4. The
diagrams show the bit packing in "network byte order," also known as diagrams show the bit packing in "network byte order," also known as
skipping to change at page 20, line 47 skipping to change at page 21, line 44
| 1 1 1| | | | | | | 1 1 1| | | | | |
|5 6 7 8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6 7| |5 6 7 8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6 7|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| C2 | S2 | GA2 | GB2 | | C2 | S2 | GA2 | GB2 |
| 1 1 1| | | | | 1 1 1| | | |
|8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3| |8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 4: G.729 and G.729A bit packing Figure 4: G.729 and G.729A bit packing
The packing of the G.729 Annex B comfort noise frame is shown in Fig. The packing of the G.729 Annex B comfort noise frame is shown in
5. Fig. 5.
0 1 0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|L| LSF1 | LSF2 | GAIN |R| |L| LSF1 | LSF2 | GAIN |R|
|S| | | |E| |S| | | |E|
|F| | | |S| |F| | | |S|
|0|0 1 2 3 4|0 1 2 3|0 1 2 3 4|V| RESV = Reserved (zero) |0|0 1 2 3 4|0 1 2 3|0 1 2 3 4|V| RESV = Reserved (zero)
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 5: G.729 Annex B bit packing Figure 5: G.729 Annex B bit packing
4.5.7 G729D and G729E 4.5.7 G729D and G729E
Annexes D and E to ITU-T Recommendation G.729 provide additional data Annexes D and E to ITU-T Recommendation G.729 provide additional data
rates. Because the data rate is not signaled in the bitstream, the rates. Because the data rate is not signaled in the bitstream, the
different data rates are given distinct RTP encoding names which are different data rates are given distinct RTP encoding names which are
mapped to distinct payload type numbers. G729D indicates a 6.4 kbit/s mapped to distinct payload type numbers. G729D indicates a 6.4
coding mode (G.729 Annex D, for momentary reduction in channel kbit/s coding mode (G.729 Annex D, for momentary reduction in channel
capacity), while G729E indicates an 11.8 kbit/s mode (G.729 Annex E, capacity), while G729E indicates an 11.8 kbit/s mode (G.729 Annex E,
for improved performance with a wide range of narrow-band input for improved performance with a wide range of narrow-band input
signals, e.g. music and background noise). Annex E has two operating signals, e.g. music and background noise). Annex E has two operating
modes, backward adaptive and forward adaptive, which are signaled by modes, backward adaptive and forward adaptive, which are signaled by
the first two bits in each frame (the most significant two bits of the first two bits in each frame (the most significant two bits of
the first octet). the first octet).
The voice activity detector (VAD) and comfort noise generator (CNG) The voice activity detector (VAD) and comfort noise generator (CNG)
algorithm specified in Annex B of G.729 may be used with Annex D and algorithm specified in Annex B of G.729 may be used with Annex D and
Annex E frames in addition to G.729 and G.729 Annex A frames. The Annex E frames in addition to G.729 and G.729 Annex A frames. The
algorithm details for the operation of Annexes D and E with the Annex algorithm details for the operation of Annexes D and E with the Annex
B CNG are specified in G.729 Annexes F and G. Note that Annexes F and B CNG are specified in G.729 Annexes F and G. Note that Annexes F
G do not introduce any new encodings. The MIME registrations for and G do not introduce any new encodings. Receivers MUST accept
G729D and G729E in RFC YYYY [7] specify a parameter that MAY be used comfort noise frames if restriction of their use has not been
with MIME or SDP to restrict the use of comfort noise frames. signaled. The MIME registrations for G729D and G729E in RFC YYYY [7]
specify a parameter that MAY be used with MIME or SDP to restrict the
use of comfort noise frames.
For G729D, an RTP packet may consist of zero or more G.729 Annex D For G729D, an RTP packet may consist of zero or more G.729 Annex D
frames, followed by zero or one G.729 Annex B frame. Similarly, for frames, followed by zero or one G.729 Annex B frame. Similarly, for
G729E, an RTP packet may consist of zero or more G.729 Annex E G729E, an RTP packet may consist of zero or more G.729 Annex E
frames, followed by zero or one G.729 Annex B frame. The presence of frames, followed by zero or one G.729 Annex B frame. The presence of
a comfort noise frame can be deduced from the length of the RTP a comfort noise frame can be deduced from the length of the RTP
payload. payload.
A single RTP packet must contain frames of only one data rate, A single RTP packet must contain frames of only one data rate,
optionally followed by one comfort noise frame. The data rate may be optionally followed by one comfort noise frame. The data rate may be
skipping to change at page 24, line 32 skipping to change at page 25, line 32
|6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1| |6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 8: G.729 Annex E (backward adaptive mode) bit packing Figure 8: G.729 Annex E (backward adaptive mode) bit packing
4.5.8 GSM 4.5.8 GSM
GSM (group speciale mobile) denotes the European GSM 06.10 standard GSM (group speciale mobile) denotes the European GSM 06.10 standard
for full-rate speech transcoding, ETS 300 961, which is based on for full-rate speech transcoding, ETS 300 961, which is based on
RPE/LTP (residual pulse excitation/long term prediction) coding at a RPE/LTP (residual pulse excitation/long term prediction) coding at a
rate of 13 kb/s [11,12,13]. The text of the standard can be obtained rate of 13 kb/s [12,13,14]. The text of the standard can be obtained
from from
ETSI (European Telecommunications Standards Institute) ETSI (European Telecommunications Standards Institute)
ETSI Secretariat: B.P.152 ETSI Secretariat: B.P.152
F-06561 Valbonne Cedex F-06561 Valbonne Cedex
France France
Phone: +33 92 94 42 00 Phone: +33 92 94 42 00
Fax: +33 93 65 47 16 Fax: +33 93 65 47 16
Blocks of 160 audio samples are compressed into 33 octets, for an Blocks of 160 audio samples are compressed into 33 octets, for an
effective data rate of 13,200 b/s. effective data rate of 13,200 b/s.
4.5.8.1 General Packaging Issues General Packaging Issues
The GSM standard (ETS 300 961) specifies the bit stream produced by The GSM standard (ETS 300 961) specifies the bit stream produced by
the codec, but does not specify how these bits should be packed for the codec, but does not specify how these bits should be packed for
transmission. The packetization specified here has subsequently been transmission. The packetization specified here has subsequently been
adopted in ETSI Technical Specification TS 101 318. Some software adopted in ETSI Technical Specification TS 101 318. Some software
implementations of the GSM codec use a different packing than that implementations of the GSM codec use a different packing than that
specified here. specified here.
In the GSM packing used by RTP, the bits SHALL be packed beginning In the GSM packing used by RTP, the bits SHALL be packed beginning
from the most significant bit. Every 160 sample GSM frame is coded from the most significant bit. Every 160 sample GSM frame is coded
into one 33 octet (264 bit) buffer. Every such buffer begins with a 4 into one 33 octet (264 bit) buffer. Every such buffer begins with a
bit signature (0xD), followed by the MSB encoding of the fields of 4 bit signature (0xD), followed by the MSB encoding of the fields of
the frame. The first octet thus contains 1101 in the 4 most the frame. The first octet thus contains 1101 in the 4 most
significant bits (0-3) and the 4 most significant bits of F1 (0-3) in significant bits (0-3) and the 4 most significant bits of F1 (0-3) in
the 4 least significant bits (4-7). The second octet contains the 2 the 4 least significant bits (4-7). The second octet contains the 2
least significant bits of F1 in bits 0-1, and F2 in bits 2-7, and so least significant bits of F1 in bits 0-1, and F2 in bits 2-7, and so
on. The order of the fields in the frame is described in Table 2. on. The order of the fields in the frame is described in Table 2.
4.5.8.2 GSM variable names and numbers GSM variable names and numbers
In the RTP encoding we have the bit pattern described in Table 3, In the RTP encoding we have the bit pattern described in Table 3,
where F.i signifies the ith bit of the field F, bit 0 is the most where F.i signifies the ith bit of the field F, bit 0 is the most
significant bit, and the bits of every octet are numbered from 0 to 7 significant bit, and the bits of every octet are numbered from 0 to 7
from most to least significant. from most to least significant.
4.5.9 GSM-EFR 4.5.9 GSM-EFR
GSM-EFR denotes GSM 06.60 enhanced full rate speech transcoding, GSM-EFR denotes GSM 06.60 enhanced full rate speech transcoding,
specified in ETS 300 969 which is available from ETSI at the address specified in ETS 300 969 which is available from ETSI at the address
skipping to change at page 25, line 49 skipping to change at page 27, line 5
zero. zero.
4.5.11 L16 4.5.11 L16
L16 denotes uncompressed audio data samples, using 16-bit signed L16 denotes uncompressed audio data samples, using 16-bit signed
representation with 65535 equally divided steps between minimum and representation with 65535 equally divided steps between minimum and
maximum signal level, ranging from -32768 to 32767. The value is maximum signal level, ranging from -32768 to 32767. The value is
represented in two's complement notation and transmitted in network represented in two's complement notation and transmitted in network
byte order (most significant byte first). byte order (most significant byte first).
The MIME registration for L16 in RFC YYYY [7] specifies parameters
field field name bits field field name bits field field name bits field field name bits
________________________________________________ ________________________________________________
1 LARc[0] 6 39 xmc[22] 3 1 LARc[0] 6 39 xmc[22] 3
2 LARc[1] 6 40 xmc[23] 3 2 LARc[1] 6 40 xmc[23] 3
3 LARc[2] 5 41 xmc[24] 3 3 LARc[2] 5 41 xmc[24] 3
4 LARc[3] 5 42 xmc[25] 3 4 LARc[3] 5 42 xmc[25] 3
5 LARc[4] 4 43 Nc[2] 7 5 LARc[4] 4 43 Nc[2] 7
6 LARc[5] 4 44 bc[2] 2 6 LARc[5] 4 44 bc[2] 2
7 LARc[6] 3 45 Mc[2] 2 7 LARc[6] 3 45 Mc[2] 2
8 LARc[7] 3 46 xmaxc[2] 6 8 LARc[7] 3 46 xmaxc[2] 6
skipping to change at page 26, line 47 skipping to change at page 27, line 48
32 xmc[15] 3 70 xmc[45] 3 32 xmc[15] 3 70 xmc[45] 3
33 xmc[16] 3 71 xmc[46] 3 33 xmc[16] 3 71 xmc[46] 3
34 xmc[17] 3 72 xmc[47] 3 34 xmc[17] 3 72 xmc[47] 3
35 xmc[18] 3 73 xmc[48] 3 35 xmc[18] 3 73 xmc[48] 3
36 xmc[19] 3 74 xmc[49] 3 36 xmc[19] 3 74 xmc[49] 3
37 xmc[20] 3 75 xmc[50] 3 37 xmc[20] 3 75 xmc[50] 3
38 xmc[21] 3 76 xmc[51] 3 38 xmc[21] 3 76 xmc[51] 3
Table 2: Ordering of GSM variables Table 2: Ordering of GSM variables
The MIME registration for L16 in RFC YYYY [7] specifies parameters
that MAY be used with MIME or SDP to indicate that analog preemphasis that MAY be used with MIME or SDP to indicate that analog preemphasis
was applied to the signal before quantization or to indicate that a was applied to the signal before quantization or to indicate that a
multiple-channel audio stream follows a different channel ordering multiple-channel audio stream follows a different channel ordering
convention than is specified in Section 4.1.
Octet Bit 0 Bit 1 Bit 2 Bit 3 Bit 4 Bit 5 Bit 6 Bit 7 Octet Bit 0 Bit 1 Bit 2 Bit 3 Bit 4 Bit 5 Bit 6 Bit 7
_____________________________________________________________________ _____________________________________________________________________
0 1 1 0 1 LARc0.0 LARc0.1 LARc0.2 LARc0.3 0 1 1 0 1 LARc0.0 LARc0.1 LARc0.2 LARc0.3
1 LARc0.4 LARc0.5 LARc1.0 LARc1.1 LARc1.2 LARc1.3 LARc1.4 LARc1.5 1 LARc0.4 LARc0.5 LARc1.0 LARc1.1 LARc1.2 LARc1.3 LARc1.4 LARc1.5
2 LARc2.0 LARc2.1 LARc2.2 LARc2.3 LARc2.4 LARc3.0 LARc3.1 LARc3.2 2 LARc2.0 LARc2.1 LARc2.2 LARc2.3 LARc2.4 LARc3.0 LARc3.1 LARc3.2
3 LARc3.3 LARc3.4 LARc4.0 LARc4.1 LARc4.2 LARc4.3 LARc5.0 LARc5.1 3 LARc3.3 LARc3.4 LARc4.0 LARc4.1 LARc4.2 LARc4.3 LARc5.0 LARc5.1
4 LARc5.2 LARc5.3 LARc6.0 LARc6.1 LARc6.2 LARc7.0 LARc7.1 LARc7.2 4 LARc5.2 LARc5.3 LARc6.0 LARc6.1 LARc6.2 LARc7.0 LARc7.1 LARc7.2
5 Nc0.0 Nc0.1 Nc0.2 Nc0.3 Nc0.4 Nc0.5 Nc0.6 bc0.0 5 Nc0.0 Nc0.1 Nc0.2 Nc0.3 Nc0.4 Nc0.5 Nc0.6 bc0.0
6 bc0.1 Mc0.0 Mc0.1 xmaxc00 xmaxc01 xmaxc02 xmaxc03 xmaxc04 6 bc0.1 Mc0.0 Mc0.1 xmaxc00 xmaxc01 xmaxc02 xmaxc03 xmaxc04
7 xmaxc05 xmc0.0 xmc0.1 xmc0.2 xmc1.0 xmc1.1 xmc1.2 xmc2.0 7 xmaxc05 xmc0.0 xmc0.1 xmc0.2 xmc1.0 xmc1.1 xmc1.2 xmc2.0
skipping to change at page 27, line 43 skipping to change at page 28, line 42
26 Nc3.0 Nc3.1 Nc3.2 Nc3.3 Nc3.4 Nc3.5 Nc3.6 bc3.0 26 Nc3.0 Nc3.1 Nc3.2 Nc3.3 Nc3.4 Nc3.5 Nc3.6 bc3.0
27 bc3.1 Mc3.0 Mc3.1 xmaxc30 xmaxc31 xmaxc32 xmaxc33 xmaxc34 27 bc3.1 Mc3.0 Mc3.1 xmaxc30 xmaxc31 xmaxc32 xmaxc33 xmaxc34
28 xmaxc35 xmc39.0 xmc39.1 xmc39.2 xmc40.0 xmc40.1 xmc40.2 xmc41.0 28 xmaxc35 xmc39.0 xmc39.1 xmc39.2 xmc40.0 xmc40.1 xmc40.2 xmc41.0
29 xmc41.1 xmc41.2 xmc42.0 xmc42.1 xmc42.2 xmc43.0 xmc43.1 xmc43.2 29 xmc41.1 xmc41.2 xmc42.0 xmc42.1 xmc42.2 xmc43.0 xmc43.1 xmc43.2
30 xmc44.0 xmc44.1 xmc44.2 xmc45.0 xmc45.1 xmc45.2 xmc46.0 xmc46.1 30 xmc44.0 xmc44.1 xmc44.2 xmc45.0 xmc45.1 xmc45.2 xmc46.0 xmc46.1
31 xmc46.2 xmc47.0 xmc47.1 xmc47.2 xmc48.0 xmc48.1 xmc48.2 xmc49.0 31 xmc46.2 xmc47.0 xmc47.1 xmc47.2 xmc48.0 xmc48.1 xmc48.2 xmc49.0
32 xmc49.1 xmc49.2 xmc50.0 xmc50.1 xmc50.2 xmc51.0 xmc51.1 xmc51.2 32 xmc49.1 xmc49.2 xmc50.0 xmc50.1 xmc50.2 xmc51.0 xmc51.1 xmc51.2
Table 3: GSM payload format Table 3: GSM payload format
convention than is specified in Section 4.1.
4.5.12 LPC 4.5.12 LPC
LPC designates an experimental linear predictive encoding contributed LPC designates an experimental linear predictive encoding contributed
by Ron Frederick, which is based on an implementation written by Ron by Ron Frederick, which is based on an implementation written by Ron
Zuckerman posted to the Usenet group comp.dsp on June 26, 1992. The Zuckerman posted to the Usenet group comp.dsp on June 26, 1992. The
codec generates 14 octets for every frame. The framesize is set to 20 codec generates 14 octets for every frame. The framesize is set to
ms, resulting in a bit rate of 5,600 b/s. 20 ms, resulting in a bit rate of 5,600 b/s.
4.5.13 MPA 4.5.13 MPA
MPA denotes MPEG-1 or MPEG-2 audio encapsulated as elementary MPA denotes MPEG-1 or MPEG-2 audio encapsulated as elementary
streams. The encoding is defined in ISO standards ISO/IEC 11172-3 streams. The encoding is defined in ISO standards ISO/IEC 11172-3
and 13818-3. The encapsulation is specified in RFC 2250 [14]. and 13818-3. The encapsulation is specified in RFC 2250 [15].
The encoding may be at any of three levels of complexity, called The encoding may be at any of three levels of complexity, called
Layer I, II and III. The selected layer as well as the sampling rate Layer I, II and III. The selected layer as well as the sampling rate
and channel count are indicated in the payload. The RTP timestamp and channel count are indicated in the payload. The RTP timestamp
clock rate is always 90000, independent of the sampling rate. MPEG-1 clock rate is always 90000, independent of the sampling rate. MPEG-1
audio supports sampling rates of 32, 44.1, and 48 kHz (ISO/IEC audio supports sampling rates of 32, 44.1, and 48 kHz (ISO/IEC
11172-3, section 1.1; "Scope"). MPEG-2 supports sampling rates of 16, 11172-3, section 1.1; "Scope"). MPEG-2 supports sampling rates of
22.05 and 24 kHz. The number of samples per frame is fixed, but the 16, 22.05 and 24 kHz. The number of samples per frame is fixed, but
frame size will vary with the sampling rate and bit rate. the frame size will vary with the sampling rate and bit rate.
The MIME registration for MPA in RFC YYYY [7] specifies parameters The MIME registration for MPA in RFC YYYY [7] specifies parameters
that MAY be used with MIME or SDP to restrict the selection of layer, that MAY be used with MIME or SDP to restrict the selection of layer,
channel count, sampling rate, and bit rate. channel count, sampling rate, and bit rate.
4.5.14 PCMA and PCMU 4.5.14 PCMA and PCMU
PCMA and PCMU are specified in ITU-T Recommendation G.711. Audio data PCMA and PCMU are specified in ITU-T Recommendation G.711. Audio
is encoded as eight bits per sample, after logarithmic scaling. PCMU data is encoded as eight bits per sample, after logarithmic scaling.
denotes mu-law scaling, PCMA A-law scaling. A detailed description is PCMU denotes mu-law scaling, PCMA A-law scaling. A detailed
given by Jayant and Noll [15]. Each G.711 octet SHALL be octet- description is given by Jayant and Noll [16]. Each G.711 octet SHALL
aligned in an RTP packet. The sign bit of each G.711 octet SHALL be octet-aligned in an RTP packet. The sign bit of each G.711 octet
correspond to the most significant bit of the octet in the RTP packet SHALL correspond to the most significant bit of the octet in the RTP
(i.e., assuming the G.711 samples are handled as octets on the host packet (i.e., assuming the G.711 samples are handled as octets on the
machine, the sign bit SHALL be the most significant bit of the octet host machine, the sign bit SHALL be the most significant bit of the
as defined by the host machine format). The 56 kb/s and 48 kb/s modes octet as defined by the host machine format). The 56 kb/s and 48
of G.711 are not applicable to RTP, since PCMA and PCMU MUST always kb/s modes of G.711 are not applicable to RTP, since PCMA and PCMU
be transmitted as 8-bit samples. MUST always be transmitted as 8-bit samples.
See Section 4.1 regarding silence suppression.
4.5.15 QCELP 4.5.15 QCELP
The Electronic Industries Association (EIA) & Telecommunications The Electronic Industries Association (EIA) & Telecommunications
Industry Association (TIA) standard IS-733, "TR45: High Rate Speech Industry Association (TIA) standard IS-733, "TR45: High Rate Speech
Service Option for Wideband Spread Spectrum Communications Systems," Service Option for Wideband Spread Spectrum Communications Systems,"
defines the QCELP audio compression algorithm for use in wireless defines the QCELP audio compression algorithm for use in wireless
CDMA applications. The QCELP CODEC compresses each 20 milliseconds of CDMA applications. The QCELP CODEC compresses each 20 milliseconds
8000 Hz, 16- bit sampled input speech into one of four different size of 8000 Hz, 16- bit sampled input speech into one of four different
output frames: Rate 1 (266 bits), Rate 1/2 (124 bits), Rate 1/4 (54 size output frames: Rate 1 (266 bits), Rate 1/2 (124 bits), Rate 1/4
bits) or Rate 1/8 (20 bits). For typical speech patterns, this (54 bits) or Rate 1/8 (20 bits). For typical speech patterns, this
results in an average output of 6.8 k bits/sec for normal mode and results in an average output of 6.8 k bits/sec for normal mode and
4.7 k bits/sec for reduced rate mode. The packetization of the QCELP 4.7 k bits/sec for reduced rate mode. The packetization of the QCELP
audio codec is described in [16]. audio codec is described in [17].
4.5.16 RED 4.5.16 RED
The redundant audio payload format "RED" is specified by RFC 2198
[17]. It defines a means by which multiple redundant copies of an The redundant audio payload format "RED" is specified by
audio packet may be transmitted in a single RTP stream. Each packet RFC 2198 [18]. It defines a means by which multiple redundant copies
in such a stream contains, in addition to the audio data for that of an audio packet may be transmitted in a single RTP stream. Each
packetization interval, a (more heavily compressed) copy of the data packet in such a stream contains, in addition to the audio data for
from a previous packetization interval. This allows an approximation that packetization interval, a (more heavily compressed) copy of the
of the data from lost packets to be recovered upon decoding of a data from a previous packetization interval. This allows an
subsequent packet, giving much improved sound quality when compared approximation of the data from lost packets to be recovered upon
with silence substitution for lost packets. decoding of a subsequent packet, giving much improved sound quality
when compared with silence substitution for lost packets.
4.5.17 VDVI 4.5.17 VDVI
VDVI is a variable-rate version of DVI4, yielding speech bit rates of VDVI is a variable-rate version of DVI4, yielding speech bit rates of
between 10 and 25 kb/s. It is specified for single-channel operation between 10 and 25 kb/s. It is specified for single-channel operation
only. Samples are packed into octets starting at the most- only. Samples are packed into octets starting at the most-
significant bit. The last octet is padded with 1 bits if the last significant bit. The last octet is padded with 1 bits if the last
sample does not fill the last octet. This padding is distinct from sample does not fill the last octet. This padding is distinct from
the valid codewords. The receiver needs to detect the padding the valid codewords. The receiver needs to detect the padding
because there is no explicit count of samples in the packet. because there is no explicit count of samples in the packet.
skipping to change at page 30, line 21 skipping to change at page 31, line 23
MAY be used. However, it is not sufficient to use the video frame MAY be used. However, it is not sufficient to use the video frame
rate (typically between 15 and 30 Hz) because that does not provide rate (typically between 15 and 30 Hz) because that does not provide
adequate resolution for typical synchronization requirements when adequate resolution for typical synchronization requirements when
calculating the RTP timestamp corresponding to the NTP timestamp in calculating the RTP timestamp corresponding to the NTP timestamp in
an RTCP SR packet. The timestamp resolution MUST also be sufficient an RTCP SR packet. The timestamp resolution MUST also be sufficient
for the jitter estimate contained in the receiver reports. for the jitter estimate contained in the receiver reports.
For most of these video encodings, the RTP timestamp encodes the For most of these video encodings, the RTP timestamp encodes the
sampling instant of the video image contained in the RTP data packet. sampling instant of the video image contained in the RTP data packet.
If a video image occupies more than one packet, the timestamp is the If a video image occupies more than one packet, the timestamp is the
same on all of those packets. Packets from different video images are same on all of those packets. Packets from different video images
distinguished by their different timestamps. are distinguished by their different timestamps.
Most of these video encodings also specify that the marker bit of the Most of these video encodings also specify that the marker bit of the
RTP header SHOULD be set to one in the last packet of a video frame RTP header SHOULD be set to one in the last packet of a video frame
and otherwise set to zero. Thus, it is not necessary to wait for a and otherwise set to zero. Thus, it is not necessary to wait for a
following packet with a different timestamp to detect that a new following packet with a different timestamp to detect that a new
frame should be displayed. frame should be displayed.
5.1 CelB 5.1 CelB
The CELL-B encoding is a proprietary encoding proposed by Sun The CELL-B encoding is a proprietary encoding proposed by Sun
Microsystems. The byte stream format is described in RFC 2029 [18]. Microsystems. The byte stream format is described in RFC 2029 [19].
5.2 JPEG 5.2 JPEG
The encoding is specified in ISO Standards 10918-1 and 10918-2. The The encoding is specified in ISO Standards 10918-1 and 10918-2. The
RTP payload format is as specified in RFC 2435 [19]. RTP payload format is as specified in RFC 2435 [20].
5.3 H261 5.3 H261
The encoding is specified in ITU-T Recommendation H.261, "Video codec The encoding is specified in ITU-T Recommendation H.261, "Video codec
for audiovisual services at p x 64 kbit/s". The packetization and for audiovisual services at p x 64 kbit/s". The packetization and
RTP-specific properties are described in RFC 2032 [20]. RTP-specific properties are described in RFC 2032 [21].
5.4 H263 5.4 H263
The encoding is specified in the 1996 version of ITU-T Recommendation The encoding is specified in the 1996 version of ITU-T Recommendation
H.263, "Video coding for low bit rate communication". The H.263, "Video coding for low bit rate communication". The
packetization and RTP-specific properties are described in RFC 2190 packetization and RTP-specific properties are described in
[21]. The H263-1998 payload format is RECOMMENDED over this one for RFC 2190 [22]. The H263-1998 payload format is RECOMMENDED over this
use by new implementations. one for use by new implementations.
5.5 H263-1998 5.5 H263-1998
The encoding is specified in the 1998 version of ITU-T Recommendation The encoding is specified in the 1998 version of ITU-T Recommendation
H.263, "Video coding for low bit rate communication". The H.263, "Video coding for low bit rate communication". The
packetization and RTP-specific properties are described in RFC 2429 packetization and RTP-specific properties are described in
[22]. Because the 1998 version of H.263 is a superset of the 1996 RFC 2429 [23]. Because the 1998 version of H.263 is a superset of
syntax, this payload format can also be used with the 1996 version of the 1996 syntax, this payload format can also be used with the 1996
H.263, and is RECOMMENDED for this use by new implementations. This version of H.263, and is RECOMMENDED for this use by new
payload format does not replace RFC 2190, which continues to be used implementations. This payload format does not replace RFC 2190,
by existing implementations, and may be required for backward which continues to be used by existing implementations, and may be
compatibility in new implementations. Implementations using the new required for backward compatibility in new implementations.
features of the 1998 version of H.263 MUST use the payload format Implementations using the new features of the 1998 version of H.263
described in RFC 2429. MUST use the payload format described in RFC 2429.
5.6 MPV 5.6 MPV
MPV designates the use of MPEG-1 and MPEG-2 video encoding elementary MPV designates the use of MPEG-1 and MPEG-2 video encoding elementary
streams as specified in ISO Standards ISO/IEC 11172 and 13818-2, streams as specified in ISO Standards ISO/IEC 11172 and 13818-2,
respectively. The RTP payload format is as specified in RFC 2250 respectively. The RTP payload format is as specified in
[14], Section 3. RFC 2250 [15], Section 3.
The MIME registration for MPV in RFC YYYY [7] specifies a parameter The MIME registration for MPV in RFC YYYY [7] specifies a parameter
that MAY be used with MIME or SDP to restrict the selection of the that MAY be used with MIME or SDP to restrict the selection of the
type of MPEG video. type of MPEG video.
5.7 MP2T 5.7 MP2T
MP2T designates the use of MPEG-2 transport streams, for either audio MP2T designates the use of MPEG-2 transport streams, for either audio
or video. The RTP payload format is described in RFC 2250 [14], or video. The RTP payload format is described in RFC 2250 [15],
Section 2. Section 2.
5.8 nv 5.8 nv
The encoding is implemented in the program `nv', version 4, developed The encoding is implemented in the program `nv', version 4, developed
at Xerox PARC by Ron Frederick. Further information is available from at Xerox PARC by Ron Frederick. Further information is available
the author: from the author:
Ron Frederick Ron Frederick
Cacheflow Inc. Blue Coat Systems Inc.
650 Almanor Avenue 650 Almanor Avenue
Sunnyvale, CA 94085 Sunnyvale, CA 94085
United States United States
electronic mail: ronf@cacheflow.com electronic mail: ronf@bluecoat.com
6 Payload Type Definitions 6 Payload Type Definitions
Tables 4 and 5 define this profile's static payload type values for Tables 4 and 5 define this profile's static payload type values for
the PT field of the RTP data header. In addition, payload type the PT field of the RTP data header. In addition, payload type
values in the range 96-127 MAY be defined dynamically through a values in the range 96-127 MAY be defined dynamically through a
conference control protocol, which is beyond the scope of this conference control protocol, which is beyond the scope of this
document. For example, a session directory could specify that for a document. For example, a session directory could specify that for a
given session, payload type 96 indicates PCMU encoding, 8,000 Hz given session, payload type 96 indicates PCMU encoding, 8,000 Hz
sampling rate, 2 channels. Entries in Tables 4 and 5 with payload sampling rate, 2 channels. Entries in Tables 4 and 5 with payload
type "dyn" have no static payload type assigned and are only used type "dyn" have no static payload type assigned and are only used
with a dynamic payload type. Payload type 13 is reserved for a with a dynamic payload type. Payload type 2 was assigned to G721 in
comfort noise payload format to be specified in a separate RFC. RFC 1890 and to its equivalent successor G726-32 in draft versions of
Payload type 19 is also marked "reserved" because some draft versions this specification, but its use is now deprecated and that static
of this specification assigned that number to a comfort noise payload payload type is marked reserved due to conflicting use for the
format. The payload type range 72-76 is marked "reserved" so that payload formats G726-32 and AAL2-G726-32 (see Section 4.5.4).
RTCP and RTP packets can be reliably distinguished (see Section Payload type 13 indicates the Comfort Noise (CN) payload format
"Summary of Protocol Constants" of the RTP protocol specification). specified in RFC 3389 [9]. Payload type 19 is marked "reserved"
because some draft versions of this specification assigned that
number to an earlier version of the comfort noise payload format.
The payload type range 72-76 is marked "reserved" so that RTCP and
RTP packets can be reliably distinguished (see Section "Summary of
Protocol Constants" of the RTP protocol specification).
The payload types currently defined in this profile are assigned to The payload types currently defined in this profile are assigned to
exactly one of three categories or media types : audio only, video exactly one of three categories or media types : audio only, video
only and those combining audio and video. The media types are marked only and those combining audio and video. The media types are marked
in Tables 4 and 5 as "A", "V" and "AV", respectively. Payload types in Tables 4 and 5 as "A", "V" and "AV", respectively. Payload types
of different media types SHALL NOT be interleaved or multiplexed of different media types SHALL NOT be interleaved or multiplexed
within a single RTP session, but multiple RTP sessions MAY be used in within a single RTP session, but multiple RTP sessions MAY be used in
parallel to send multiple media types. An RTP source MAY change parallel to send multiple media types. An RTP source MAY change
payload types within the same media type during a session. See the payload types within the same media type during a session. See the
section "Multiplexing RTP Sessions" of RFC XXXX for additional section "Multiplexing RTP Sessions" of RFC XXXX for additional
skipping to change at page 32, line 41 skipping to change at page 34, line 4
session. This set MAY, for example, be defined by the capabilities session. This set MAY, for example, be defined by the capabilities
of the applications used, negotiated by a conference control protocol of the applications used, negotiated by a conference control protocol
or established by agreement between the human participants. or established by agreement between the human participants.
Audio applications operating under this profile SHOULD, at a minimum, Audio applications operating under this profile SHOULD, at a minimum,
be able to send and/or receive payload types 0 (PCMU) and 5 (DVI4). be able to send and/or receive payload types 0 (PCMU) and 5 (DVI4).
This allows interoperability without format negotiation and ensures This allows interoperability without format negotiation and ensures
successful negotiation with a conference control protocol. successful negotiation with a conference control protocol.
7 RTP over TCP and Similar Byte Stream Protocols 7 RTP over TCP and Similar Byte Stream Protocols
Under special circumstances, it may be necessary to carry RTP in
protocols offering a byte stream abstraction, such as TCP, possibly
multiplexed with other data. The application MUST define its own
method of delineating RTP and RTCP packets (RTSP [23] provides an
example of such an encapsulation specification.)
8 Port Assignment
PT encoding media type clock rate channels PT encoding media type clock rate channels
name (Hz) name (Hz)
___________________________________________________ ___________________________________________________
0 PCMU A 8000 1 0 PCMU A 8000 1
1 reserved A 1 reserved A
2 G726-32 A 8000 1 2 reserved A
3 GSM A 8000 1 3 GSM A 8000 1
4 G723 A 8000 1 4 G723 A 8000 1
5 DVI4 A 8000 1 5 DVI4 A 8000 1
6 DVI4 A 16000 1 6 DVI4 A 16000 1
7 LPC A 8000 1 7 LPC A 8000 1
8 PCMA A 8000 1 8 PCMA A 8000 1
9 G722 A 8000 1 9 G722 A 8000 1
10 L16 A 44100 2 10 L16 A 44100 2
11 L16 A 44100 1 11 L16 A 44100 1
12 QCELP A 8000 1 12 QCELP A 8000 1
13 reserved A 13 CN A 8000 1
14 MPA A 90000 (see text) 14 MPA A 90000 (see text)
15 G728 A 8000 1 15 G728 A 8000 1
16 DVI4 A 11025 1 16 DVI4 A 11025 1
17 DVI4 A 22050 1 17 DVI4 A 22050 1
18 G729 A 8000 1 18 G729 A 8000 1
19 reserved A 19 reserved A
20 unassigned A 20 unassigned A
21 unassigned A 21 unassigned A
22 unassigned A 22 unassigned A
23 unassigned A 23 unassigned A
dyn G726-40 A 8000 1 dyn G726-40 A 8000 1
dyn G726-32 A 8000 1
dyn G726-24 A 8000 1 dyn G726-24 A 8000 1
dyn G726-16 A 8000 1 dyn G726-16 A 8000 1
dyn G729D A 8000 1 dyn G729D A 8000 1
dyn G729E A 8000 1 dyn G729E A 8000 1
dyn GSM-EFR A 8000 1 dyn GSM-EFR A 8000 1
dyn L8 A var. var. dyn L8 A var. var.
dyn RED A (see text) dyn RED A (see text)
dyn VDVI A var. 1 dyn VDVI A var. 1
Table 4: Payload types (PT) for audio encodings Table 4: Payload types (PT) for audio encodings
As specified in the RTP protocol definition, RTP data SHOULD be Under special circumstances, it may be necessary to carry RTP in
carried on an even UDP port number and the corresponding RTCP packets protocols offering a byte stream abstraction, such as TCP, possibly
SHOULD be carried on the next higher (odd) port number. multiplexed with other data. The application MUST define its own
method of delineating RTP and RTCP packets (RTSP [24] provides an
Applications operating under this profile MAY use any such UDP port example of such an encapsulation specification.)
pair. For example, the port pair MAY be allocated randomly by a
session management program. A single fixed port number pair cannot be
required because multiple applications using this profile are likely
PT encoding media type clock rate PT encoding media type clock rate
name (Hz) name (Hz)
____________________________________________ _____________________________________________
24 unassigned V 24 unassigned V
25 CelB V 90000 25 CelB V 90000
26 JPEG V 90000 26 JPEG V 90000
27 unassigned V 27 unassigned V
28 nv V 90000 28 nv V 90000
29 unassigned V 29 unassigned V
30 unassigned V 30 unassigned V
31 H261 V 90000 31 H261 V 90000
32 MPV V 90000 32 MPV V 90000
33 MP2T AV 90000 33 MP2T AV 90000
34 H263 V 90000 34 H263 V 90000
35-71 unassigned ? 35-71 unassigned ?
72-76 reserved N/A N/A 72-76 reserved N/A N/A
77-95 unassigned ? 77-95 unassigned ?
96-127 dynamic ? 96-127 dynamic ?
dyn H263-1998 V 90000 dyn H263-1998 V 90000
Table 5: Payload types (PT) for video and combined encodings Table 5: Payload types (PT) for video and combined encodings
to run on the same host, and there are some operating systems that do 8 Port Assignment
not allow multiple processes to use the same UDP port with different
multicast addresses. As specified in the RTP protocol definition, RTP data SHOULD be
carried on an even UDP port number and the corresponding RTCP packets
SHOULD be carried on the next higher (odd) port number.
Applications operating under this profile MAY use any such UDP port
pair. For example, the port pair MAY be allocated randomly by a
session management program. A single fixed port number pair cannot
be required because multiple applications using this profile are
likely to run on the same host, and there are some operating systems
that do not allow multiple processes to use the same UDP port with
different multicast addresses.
However, port numbers 5004 and 5005 have been registered for use with However, port numbers 5004 and 5005 have been registered for use with
this profile for those applications that choose to use them as the this profile for those applications that choose to use them as the
default pair. Applications that operate under multiple profiles MAY default pair. Applications that operate under multiple profiles MAY
use this port pair as an indication to select this profile if they use this port pair as an indication to select this profile if they
are not subject to the constraint of the previous paragraph. are not subject to the constraint of the previous paragraph.
Applications need not have a default and MAY require that the port Applications need not have a default and MAY require that the port
pair be explicitly specified. The particular port numbers were chosen pair be explicitly specified. The particular port numbers were
to lie in the range above 5000 to accommodate port number allocation chosen to lie in the range above 5000 to accommodate port number
practice within some versions of the Unix operating system, where allocation practice within some versions of the Unix operating
port numbers below 1024 can only be used by privileged processes and system, where port numbers below 1024 can only be used by privileged
port numbers between 1024 and 5000 are automatically assigned by the processes and port numbers between 1024 and 5000 are automatically
operating system. assigned by the operating system.
9 Changes from RFC 1890 A Changes from RFC 1890
This RFC revises RFC 1890. It is mostly backwards-compatible with RFC This RFC revises RFC 1890. It is mostly backwards-compatible with
1890 except for functions removed because two interoperable RFC 1890 except for functions removed because two interoperable
implementations were not found. The additions to RFC 1890 codify implementations were not found. The additions to RFC 1890 codify
existing practice in the use of payload formats under this profile. existing practice in the use of payload formats under this profile.
Since this profile may be used without using any of the payload Since this profile may be used without using any of the payload
formats listed here, the addition of new payload formats in this formats listed here, the addition of new payload formats in this
revision does not affect backwards compatibility. The changes are revision does not affect backwards compatibility. The changes are
listed below, categorized into functional and non-functional changes. listed below, categorized into functional and non-functional changes.
Functional changes: Functional changes:
o A new Section "IANA Considerations" was added to specify the o A new Appendix C "IANA Considerations" was added to specify
registration of the name for this profile and to establish a the registration of the name for this profile. That appendix
new policy that no additional registration of static payload also references a new Section 3 "Registering Additional
types for this profile will be made beyond those added in this Encodings" which establishes a policy that no additional
revision and included in Tables 4 and 5. Instead, additional registration of static payload types for this profile will be
encoding names may be registered as MIME subtypes for binding made beyond those added in this revision and included in
to dynamic payload types. Non-normative references were added Tables 4 and 5. Instead, additional encoding names may be
to RFC YYYY [7] where MIME subtypes for all the listed payload registered as MIME subtypes for binding to dynamic payload
formats are registered, some with optional parameters for use types. Non-normative references were added to RFC YYYY [7]
of the payload formats. where MIME subtypes for all the listed payload formats are
registered, some with optional parameters for use of the
payload formats.
o Static payload types 4, 16, 17 and 34 were added to o Static payload types 4, 16, 17 and 34 were added to
incorporate IANA registrations made since the publication of incorporate IANA registrations made since the publication of
RFC 1890, along with the corresponding payload format RFC 1890, along with the corresponding payload format
descriptions for G723 and H263. descriptions for G723 and H263.
o Following working group discussion, static payload types 12 o Following working group discussion, static payload types 12
and 18 were added along with the corresponding payload format and 18 were added along with the corresponding payload format
descriptions for QCELP and G729. Static payload type 13 was descriptions for QCELP and G729. Static payload type 13 was
reserved for a comfort noise payload format to be defined in a assigned to the Comfort Noise (CN) payload format defined in
separate RFC. Payload type 19 was marked reserved because it RFC 3389. Payload type 19 was marked reserved because it had
had been temporarily allocated in some draft revisions of this been temporarily allocated to an earlier version of Comfort
document. Noise present in some draft revisions of this document.
o The payload format for G721 was renamed to G726-32 following o The payload format for G721 was renamed to G726-32 following
the ITU-T renumbering. the ITU-T renumbering, and the payload format description for
G726 was expanded to include the -16, -24 and -40 data rates.
Because of confusion regarding draft revisions of this
document, some implementations of these G726 payload formats
packed samples into octets starting with the most significant
bit rather than the least significant bit as specified here.
o The payload format description for G726 was expanded to To partially resolve this incompatibility, new payload formats
include the -16, -24 and -40 data rates. Payload formats G729D named AAL2-G726-16, -24, -32 and -48 will be specified in a
and G729E were added following the ITU-T addition of Annexes D separate document (see note in Section 4.5.4), and use of
and E to Recommendation G.729. Listings were added for payload static payload type 2 is deprecated as explained in Section 6.
formats GSM-EFR, RED, and H263-1998 published in other
documents subsequent to RFC 1890. These additional payload o Payload formats G729D and G729E were added following the ITU-T
formats are referenced only by dynamic payload type numbers. addition of Annexes D and E to Recommendation G.729. Listings
were added for payload formats GSM-EFR, RED, and H263-1998
published in other documents subsequent to RFC 1890. These
additional payload formats are referenced only by dynamic
payload type numbers.
o The descriptions of the payload formats for G722, G728, GSM, o The descriptions of the payload formats for G722, G728, GSM,
VDVI were expanded. VDVI were expanded.
o The payload format for 1016 audio was removed and its static o The payload format for 1016 audio was removed and its static
payload type assignment 1 was marked "reserved" because two payload type assignment 1 was marked "reserved" because two
interoperable implementations were not found. interoperable implementations were not found.
o Requirements for congestion control were added in Section 2. o Requirements for congestion control were added in Section 2.
o This profile follows the suggestion in the revised RTP spec o This profile follows the suggestion in the revised RTP spec
that RTCP bandwidth may be specified separately from the that RTCP bandwidth may be specified separately from the
session bandwidth and separately for active senders and session bandwidth and separately for active senders and
passive receivers. passive receivers.
o The mapping of a user pass-phrase string into an encryption o The mapping of a user pass-phrase string into an encryption
key was deleted from Section 2 because two interoperable key was deleted from Section 2 because two interoperable
implementations were not found. implementations were not found.
o The "quadrophonic" sample ordering convention for four-channel
audio was removed to eliminate an ambiguity as noted in
Section 4.1.
Non-functional changes: Non-functional changes:
o In Section 4.1, it is now explicitly stated that silence
suppression is allowed for all audio payload formats. (This
has always been the case and derives from a fundamental aspect
of RTP's design and the motivations for packet audio, but was
not explicit stated before.) The use of comfort noise is also
explained.
o In Section 4.1, the requirement level for setting of the o In Section 4.1, the requirement level for setting of the
marker bit on the first packet after silence for audio was marker bit on the first packet after silence for audio was
changed from "is" to "SHOULD be", and clarified that the changed from "is" to "SHOULD be", and clarified that the
marker bit is set only when packets are intentionally not marker bit is set only when packets are intentionally not
sent. sent.
o Similarly, text was added to specify that the marker bit o Similarly, text was added to specify that the marker bit
SHOULD be set to one on the last packet of a video frame, and SHOULD be set to one on the last packet of a video frame, and
that video frames are distinguished by their timestamps. that video frames are distinguished by their timestamps.
skipping to change at page 37, line 15 skipping to change at page 38, line 46
- The explanation of how to determine the number of audio - The explanation of how to determine the number of audio
frames in a packet from the length was expanded. frames in a packet from the length was expanded.
- More description of the allocation of bandwidth to SDES - More description of the allocation of bandwidth to SDES
items is given. items is given.
- A note was added that the convention for the order of - A note was added that the convention for the order of
channels specified in Section 4.1 may be overridden by a channels specified in Section 4.1 may be overridden by a
particular encoding or payload format specification. particular encoding or payload format specification.
- The terms MUST, SHOULD, MAY, etc. are used as defined in RFC - The terms MUST, SHOULD, MAY, etc. are used as defined in
2119. RFC 2119.
o A second author for this document was added. o A second author for this document was added.
10 Security Considerations B Security Considerations
Implementations using the profile defined in this specification are Implementations using the profile defined in this specification are
subject to the security considerations discussed in the RTP subject to the security considerations discussed in the RTP
specification [2]. This profile does not specify any different specification [1]. This profile does not specify any different
security services. The primary function of this profile is to list a security services. The primary function of this profile is to list a
set of data compression encodings for audio and video media. set of data compression encodings for audio and video media.
Confidentiality of the media streams is achieved by encryption. Confidentiality of the media streams is achieved by encryption.
Because the data compression used with the payload formats described Because the data compression used with the payload formats described
in this profile is applied end-to-end, encryption may be performed in this profile is applied end-to-end, encryption may be performed
after compression so there is no conflict between the two operations. after compression so there is no conflict between the two operations.
A potential denial-of-service threat exists for data encodings using A potential denial-of-service threat exists for data encodings using
compression techniques that have non-uniform receiver-end compression techniques that have non-uniform receiver-end
skipping to change at page 37, line 46 skipping to change at page 39, line 28
into the stream which are complex to decode and cause the receiver to into the stream which are complex to decode and cause the receiver to
be overloaded. However, the encodings described in this profile do be overloaded. However, the encodings described in this profile do
not exhibit any significant non-uniformity. not exhibit any significant non-uniformity.
As with any IP-based protocol, in some circumstances a receiver may As with any IP-based protocol, in some circumstances a receiver may
be overloaded simply by the receipt of too many packets, either be overloaded simply by the receipt of too many packets, either
desired or undesired. Network-layer authentication MAY be used to desired or undesired. Network-layer authentication MAY be used to
discard packets from undesired sources, but the processing cost of discard packets from undesired sources, but the processing cost of
the authentication itself may be too high. In a multicast the authentication itself may be too high. In a multicast
environment, pruning of specific sources may be implemented in future environment, pruning of specific sources may be implemented in future
versions of IGMP [24] and in multicast routing protocols to allow a versions of IGMP [25] and in multicast routing protocols to allow a
receiver to select which sources are allowed to reach it. receiver to select which sources are allowed to reach it.
11 Full Copyright Statement C IANA Considerations
Copyright (C) The Internet Society (2001). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.
The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.
This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
12 Acknowledgments
The comments and careful review of Simao Campos, Richard Cox and AVT
Working Group participants are gratefully acknowledged. The GSM
description was adopted from the IMTC Voice over IP Forum Service
Interoperability Implementation Agreement (January 1997). Fred Burg
and Terry Lyons helped with the G.729 description.
13 Addresses of Authors The RTP specification establishes a registry of profile names for use
by higher-level control protocols, such as the Session Description
Protocol (SDP), RFC 2327 [6], to refer to transport methods. This
profile registers the name "RTP/AVP".
Henning Schulzrinne Section 3 establishes the policy that no additional registration of
Dept. of Computer Science static RTP payload types for this profile will be made beyond those
Columbia University added in this document revision and included in Tables 4 and 5. IANA
1214 Amsterdam Avenue may reference that section in declining to accept any additional
New York, NY 10027 registration requests.
USA
electronic mail: schulzrinne@cs.columbia.edu
Stephen L. Casner D References
Packet Design
2465 Latham Street
Mountain View, CA 94040
United States
electronic mail: casner@acm.org
References
Normative References Normative References
[1] S. Bradner, "Key words for use in RFCs to Indicate Requirement [1] Schulzrinne, H., S. Casner, R. Frederick, and V. Jacobson, "RTP:
Levels," RFC 2119, Internet Engineering Task Force, Mar. 1997. A Transport Protocol for Real-Time Applications," Work in
progress, revision to RFC 1889.
[2] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A [2] Bradner, S., "Key Words for Use in RFCs to Indicate Requirement
transport protocol for real-time applications," Internet Draft, Levels," BCP 14, RFC 2119, March 1997.
Internet Engineering Task Force, Feb. 1999 Work in progress, revision
to RFC 1889.
[3] Apple Computer, "Audio interchange file format AIFF-C," Aug. [3] Apple Computer, "Audio Interchange File Format AIFF-C," August
1991. (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z). 1991. (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z).
Non-Normative References Informative References
[4] R. Braden, D. Clark, S. Shenker, "Integrated Services in the [4] Braden, R., D. Clark, S. Shenker, "Integrated Services in the
Internet Architecture: an Overview," Request for Comments Internet Architecture: an Overview," RFC 1633, June 1994.
(Informational) RFC 1633, Internet Engineering Task Force, June 1994.
[5] S. Blake, D. Black, M. Carlson, E. Davies, Z. Wang, W. Weiss, "An [5] Blake, S., D. Black, M. Carlson, E. Davies, Z. Wang, W. Weiss,
Architecture for Differentiated Service," Request for Comments "An Architecture for Differentiated Service," RFC 2475, December
(Proposed Standard) RFC 2475, Internet Engineering Task Force, Dec.
1998. 1998.
[6] M. Handley and V. Jacobson, "SDP: Session Description Protocol," [6] Handley, M. and V. Jacobson, "SDP: Session Description
Request for Comments (Proposed Standard) RFC 2327, Internet Protocol," RFC 2327, April 1998.
Engineering Task Force, Apr. 1998.
[7] S. Casner and P. Hoschka, "MIME Type Registration of RTP Payload [7] Casner, S. and P. Hoschka, "MIME Type Registration of RTP
Types," Internet Draft, Internet Engineering Task Force, July 2001. Payload Types," Internet Draft, Internet Engineering Task Force,
Work in progress. Work in progress.
[8] N. Freed, J. Klensin, and J. Postel, "Multipurpose Internet Mail [8] Freed, N., J. Klensin, and J. Postel, "Multipurpose Internet
Extensions (MIME) Part Four: Registration Procedures," RFC 2048, Mail Extensions (MIME) Part Four: Registration Procedures,"
Internet Engineering Task Force, Nov. 1996. BCP 13, RFC 2048, November 1996.
[9] IMA Digital Audio Focus and Technical Working Groups,
"Recommended practices for enhancing digital audio compatibility in
multimedia systems (version 3.00)," tech. rep., Interactive
Multimedia Association, Annapolis, Maryland, Oct. 1992.
[10] D. Deleam and J.-P. Petit, "Real-time implementations of the
recent ITU-T low bit rate speech coders on the TI TMS320C54X DSP:
results, methodology, and applications," in Proc. of International
Conference on Signal Processing, Technology, and Applications
(ICSPAT) , (Boston, Massachusetts), pp. 1656--1660, Oct. 1996.
[11] M. Mouly and M.-B. Pautet, The GSM system for mobile [9] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
communications Lassay-les-Chateaux, France: Europe Media Duplication, Comfort Noise (CN)," RFC 3389, September 2002.
1993.
[12] J. Degener, "Digital speech compression," Dr. Dobb's Journal , [10] IMA Digital Audio Focus and Technical Working Groups,
Dec. 1994. "Recommended practices for enhancing digital audio compatibility
in multimedia systems (version 3.00)," tech. rep., Interactive
Multimedia Association, Annapolis, Maryland, October 1992.
[13] S. M. Redl, M. K. Weber, and M. W. Oliphant, An Introduction to [11] Deleam, D. and J.-P. Petit, "Real-time implementations of the
GSM Boston: Artech House, 1995. recent ITU-T low bit rate speech coders on the TI TMS320C54X
DSP: results, methodology, and applications," in Proc. of
International Conference on Signal Processing, Technology, and
Applications (ICSPAT) , (Boston, Massachusetts), pp. 1656--1660,
October 1996.
[14] D. Hoffman, G. Fernando, V. Goyal, and M. Civanlar, "RTP payload [12] Mouly, M. and M.-B. Pautet, The GSM system for mobile
format for MPEG1/MPEG2 video," Request for Comments (Proposed communications Lassay-les-Chateaux, France: Europe Media
Standard) RFC 2250, Internet Engineering Task Force, Jan. 1998. Duplication, 1993.
[15] N. S. Jayant and P. Noll, Digital Coding of Waveforms-- [13] Degener, J., "Digital Speech Compression," Dr. Dobb's Journal ,
Principles and Applications to Speech and Video Englewood Cliffs, New December 1994.
Jersey: Prentice-Hall, 1984.
[16] K. McKay, "RTP Payload Format for PureVoice(tm) Audio", Request [14] Redl, S., M. Weber, and M. Oliphant, An Introduction to GSM
for Comments (Proposed Standard) RFC 2658, Internet Engineering Task Boston: Artech House, 1995.
Force, Aug. 1999.
[17] C. Perkins, I. Kouvelas, O. Hodson, V. Hardman, M. Handley, J.C. [15] Hoffman, D., G. Fernando, V. Goyal, and M. Civanlar, "RTP
Bolot, A. Vega-Garcia, and S. Fosse-Parisis, "RTP Payload for Payload Format for MPEG1/MPEG2 Video," RFC 2250, January 1998.
Redundant Audio Data," Request for Comments (Proposed Standard) RFC
2198, Internet Engineering Task Force, Sep. 1997.
[18] M. Speer and D. Hoffman, "RTP payload format of sun's CellB [16] Jayant, N. and P. Noll, Digital Coding of Waveforms--Principles
video encoding," Request for Comments (Proposed Standard) RFC 2029, and Applications to Speech and Video Englewood Cliffs, New
Internet Engineering Task Force, Oct. 1996. Jersey: Prentice-Hall, 1984.
[19] L. Berc, W. Fenner, R. Frederick, and S. McCanne, "RTP payload [17] McKay, K., "RTP Payload Format for PureVoice(tm) Audio",
format for JPEG-compressed video," Request for Comments (Proposed RFC 2658, August 1999.
Standard) RFC 2435, Internet Engineering Task Force, Oct. 1996.
[20] T. Turletti and C. Huitema, "RTP payload format for H.261 video [18] Perkins, C., I. Kouvelas, O. Hodson, V. Hardman, M. Handley,
streams," Request for Comments (Proposed Standard) RFC 2032, Internet J.-C. Bolot, A. Vega-Garcia, and S. Fosse-Parisis, "RTP Payload
Engineering Task Force, Oct. 1996. for Redundant Audio Data," RFC 2198, September 1997.
[21] C. Zhu, "RTP payload format for H.263 video streams," Request [19] Speer, M. and D. Hoffman, "RTP Payload Format of Sun's CellB
for Comments (Proposed Standard) RFC 2190, Internet Engineering Task Video Encoding," RFC 2029, October 1996.
Force, Sep. 1997.
[22] C. Bormann, L. Cline, G. Deisher, T. Gardos, C. Maciocco, D. [20] Berc, L., W. Fenner, R. Frederick, and S. McCanne, "RTP Payload
Newell, J. Ott, G. Sullivan, S. Wenger, C. Zhu, "RTP Payload Format Format for JPEG-Compressed Video," RFC 2435, October 1996.
for the 1998 Version of ITU-T Rec. H.263 Video (H.263+)," Request for
Comments (Proposed Standard) RFC 2429, Internet Engineering Task
Force, Oct. 1998.
[23] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming [21] Turletti, T. and C. Huitema, "RTP Payload Format for H.261 Video
protocol (RTSP)," Request for Comments (Proposed Standard) RFC 2326, Streams," RFC 2032, October 1996.
Internet Engineering Task Force, Apr. 1998.
[24] S. Deering, "Host Extensions for IP Multicasting," Request for [22] Zhu, C., "RTP Payload Format for H.263 Video Streams," RFC 2190,
Comments RFC 1112, STD 5, Internet Engineering Task Force, Aug. 1989. September 1997.
Current Locations of Related Resources [23] Bormann, C., L. Cline, G. Deisher, T. Gardos, C. Maciocco,
D. Newell, J. Ott, G. Sullivan, S. Wenger, C. Zhu, "RTP Payload
Format for the 1998 Version of ITU-T Rec. H.263 Video (H.263+),"
RFC 2429, October 1998.
Note: Several sections below refer to the ITU-T Software Tool Library [24] Schulzrinne, H., A. Rao, and R. Lanphier, "Real Time Streaming
(STL). It is available from the ITU Sales Service, Place des Nations, Protocol (RTSP)," RFC 2326, April 1998.
CH-1211 Geneve 20, Switzerland (also check http://www.itu.int. The
ITU-T STL is covered by a license defined in ITU-T Recommendation
G.191, "Software tools for speech and audio coding standardization".
UTF-8 [25] Deering, S., "Host Extensions for IP Multicasting," STD 5,
RFC 1112, August 1989.
Information on the UCS Transformation Format 8 (UTF-8) is available E Current Locations of Related Resources
at
http://www.stonehand.com/unicode/standard/utf8.html Note: Several sections below refer to the ITU-T Software Tool
Library (STL). It is available from the ITU Sales Service, Place des
Nations, CH-1211 Geneve 20, Switzerland (also check
http://www.itu.int. The ITU-T STL is covered by a license defined in
ITU-T Recommendation G.191, "Software tools for speech and audio
coding standardization".
DVI4 DVI4
An implementation is available from Jack Jansen at An implementation is available from Jack Jansen at
ftp://ftp.cwi.nl/local/pub/audio/adpcm.shar ftp://ftp.cwi.nl/local/pub/audio/adpcm.shar
G722 G722
An implementation of the G.722 algorithm is available as part of the An implementation of the G.722 algorithm is available as part of the
skipping to change at page 42, line 26 skipping to change at page 42, line 44
The reference C code implementation defining the G.729 algorithm and The reference C code implementation defining the G.729 algorithm and
its Annexes A through I are available as an integral part of its Annexes A through I are available as an integral part of
Recommendation G.729 from the ITU Sales Service, listed above. Annex Recommendation G.729 from the ITU Sales Service, listed above. Annex
I contains the integrated C source code for all G.729 operating I contains the integrated C source code for all G.729 operating
modes. The G.729 algorithm and associated C code are covered by a modes. The G.729 algorithm and associated C code are covered by a
specific license. The contact information for obtaining the license specific license. The contact information for obtaining the license
is available from the ITU-T Secretariat. is available from the ITU-T Secretariat.
GSM GSM
A reference implementation was written by Carsten Borman and Jutta A reference implementation was written by Carsten Bormann and Jutta
Degener (TU Berlin, Germany). It is available at Degener (then at TU Berlin, Germany). It is available at
ftp://ftp.cs.tu-berlin.de/pub/local/kbs/tubmik/gsm/
http://www.dmn.tzi.org/software/gsm/
Although the RPE-LTP algorithm is not an ITU-T standard, there is a C Although the RPE-LTP algorithm is not an ITU-T standard, there is a C
code implementation of the RPE-LTP algorithm available as part of the code implementation of the RPE-LTP algorithm available as part of the
ITU-T STL. The STL implementation is an adaptation of the TU Berlin ITU-T STL. The STL implementation is an adaptation of the TU Berlin
version. version.
LPC LPC
An implementation is available at An implementation is available at
ftp://parcftp.xerox.com/pub/net-research/lpc.tar.Z ftp://parcftp.xerox.com/pub/net-research/lpc.tar.Z
PCMU, PCMA PCMU, PCMA
An implementation of these algorithm is available as part of the An implementation of these algorithm is available as part of the
ITU-T STL, described above. Code to convert between linear and mu-law ITU-T STL, described above. Code to convert between linear and mu-
companded data is also available in [9]. law companded data is also available in [10].
F Acknowledgments
The comments and careful review of Simao Campos, Richard Cox and AVT
Working Group participants are gratefully acknowledged. The GSM
description was adopted from the IMTC Voice over IP Forum Service
Interoperability Implementation Agreement (January 1997). Fred Burg
and Terry Lyons helped with the G.729 description.
G Addresses of Authors
Henning Schulzrinne
Department of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
electronic mail: schulzrinne@cs.columbia.edu
Stephen L. Casner
Packet Design
3400 Hillview Avenue, Building 3
Palo Alto, CA 94304
United States
electronic mail: casner@acm.org
H Intellectual Property Rights Statement
The IETF takes no position regarding the validity or scope of any
intellectual property or other rights that might be claimed to
pertain to the implementation or use of the technology described in
this document or the extent to which any license under such rights
might or might not be available; neither does it represent that it
has made any effort to identify any such rights. Information on the
IETF's procedures with respect to rights in standards-track and
standards-related documentation can be found in BCP-11. Copies of
claims of rights made available for publication and any assurances of
licenses to be made available, or the result of an attempt made to
obtain a general license or permission for the use of such
proprietary rights by implementors or users of this specification can
be obtained from the IETF Secretariat.
The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights which may cover technology that may be required to practice
this standard. Please address the information to the IETF Executive
Director.
I Full Copyright Statement
Copyright (C) The Internet Society (2003). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.
The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.
This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
 End of changes. 

This html diff was produced by rfcdiff 1.25, available from http://www.levkowetz.com/ietf/tools/rfcdiff/