draft-ietf-avt-profile-new-05.txt   draft-ietf-avt-profile-new-06.txt 
Internet Engineering Task Force AVT WG Internet Engineering Task Force AVT WG
Internet Draft Schulzrinne Internet Draft Schulzrinne/Casner
ietf-avt-profile-new-05.txt Columbia U. draft-ietf-avt-profile-new-06.txt Columbia U./Cisco Systems
February 26, 1999 June 25, 1999
Expires: August 26, 1999 Expires: December 25, 1999
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 in full conformance with
all provisions of Section 10 of RFC2026. all provisions of Section 10 of RFC2026.
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/ietf/1id-abstracts.txt
The list of Internet-Draft Shadow Directories can be accessed at 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 This memorandum is a revision of RFC 1890 in preparation for
for advancement from Proposed Standard to Draft Standard advancement from Proposed Standard to Draft Standard status. Readers
status. Readers are encouraged to use the PostScript form are encouraged to use the PostScript form of this draft to see where
of this draft to see where changes from RFC 1890 are changes from RFC 1890 are marked by change bars.
marked by change bars.
This document describes a profile called "RTP/AVP" for This document describes a profile called "RTP/AVP" for the use of the
the use of the real-time transport protocol (RTP), real-time transport protocol (RTP), version 2, and the associated
version 2, and the associated control protocol, RTCP, control protocol, RTCP, within audio and video multiparticipant
within audio and video multiparticipant conferences with conferences with minimal control. It provides interpretations of
minimal control. It provides interpretations of generic generic fields within the RTP specification suitable for audio and
fields within the RTP specification suitable for audio video conferences. In particular, this document defines a set of
and video conferences. In particular, this document default mappings from payload type numbers to encodings.
defines a set of default mappings from payload type
numbers to encodings.
This document also describes how audio and video data may This document also describes how audio and video data may be carried
be carried within RTP. It defines a set of standard within RTP. It defines a set of standard encodings and their names
encodings and their names when used within RTP. The when used within RTP. The descriptions provide pointers to reference
descriptions provide pointers to reference implementations and the detailed standards. This document is meant as
implementations and the detailed standards. This document an aid for implementors of audio, video and other real-time
is meant as an aid for implementors of audio, video and multimedia applications.
other real-time multimedia applications.
Resolution of Open Issues Resolution of Open Issues
[Note to the RFC Editor: This section is to be deleted when this [Note to the RFC Editor: This section is to be deleted when this
draft is published as an RFC but is shown here for reference during draft is published as an RFC but is shown here for reference during
the Last Call. All RFC XXXX should be filled in with the number of the Last Call. The first paragraph of the Abstract is also to be
the RTP specification RFC submitted for Draft Standard status, and deleted. All RFC XXXX should be filled in with the number of the RTP
all RFC YYYY should be filled in with the number of the draft specification RFC submitted for Draft Standard status, and all RFC
specifying MIME registration of RTP payload types as it is submitted YYYY should be filled in with the number of the draft specifying MIME
for Proposed Standard status.] registration of RTP payload types as it is submitted for Proposed
Standard status. These latter references are intended to be non-
normative.]
Readers are directed to Appendix 9, Changes from RFC 1890, for a Readers are directed to Appendix 9, Changes from RFC 1890, for a
listing of the changes that have been made in this draft. The listing of the changes that have been made in this draft. The
changes from RFC 1890 are marked with change bars in the PostScript changes from RFC 1890 are marked with change bars in the PostScript
form of this draft. form of this draft.
The revisions in this draft are intended to be complete for Working The revisions in this draft are intended to be complete for Last
Group last call. The following open issues from previous drafts have Call. The following open issues from previous drafts have been
been addressed: addressed:
o The procedure for registering encoding names as MIME subtypes o The procedure for registering RTP encoding names as MIME
is outlined here and referenced in a separate RFC-to-be that subtypes was moved to a separate RFC-to-be that may also serve
may also serve to specify how (some of) the encodings here may to specify how (some of) the encodings here may be used with
be used with mail and other not-RTP transports. mail and other not-RTP transports. That procedure is not
required to implement this profile, but may be used in those
contexts where it is needed.
o This profile follows the suggestion in the RTP spec that RTCP o This profile follows the suggestion in the RTP spec that RTCP
bandwidth may be specified separately from the session bandwidth may be specified separately from the session
bandwidth and separately for active senders and passive bandwidth and separately for active senders and passive
receivers. receivers.
o No specific action is taken in this document to address o No specific action is taken in this document to address
generic payload formats; it is assumed that if any generic generic payload formats; it is assumed that if any generic
payload formats are developed, they can be specified in payload formats are developed, they can be specified in
separate RFCs and that the session parameters they require for separate RFCs and that the session parameters they require for
skipping to change at page 3, line 45 skipping to change at page 3, line 45
(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 RFC
XXXX enumerates a number of items that can be specified or modified XXXX enumerates a number of items that can be specified or modified
in a profile. This section addresses these items. Generally, this in a profile. This section addresses these items. Generally, this
profile follows the default and/or recommended aspects of the RTP profile follows the default and/or recommended aspects of the RTP
specification. specification.
RTP data header: The standard format of the fixed RTP data header is RTP data header: The standard format of the fixed RTP data
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 appended to RTP data header additions: No additional fixed fields are
the RTP data header. appended to the RTP data header.
RTP data header extensions: No RTP header extensions are defined, but RTP data header extensions: No RTP header extensions are
applications operating under this profile MAY use such defined, but applications operating under this profile MAY
extensions. Thus, applications SHOULD NOT assume that the RTP use such extensions. Thus, applications SHOULD NOT assume
header X bit is always zero and SHOULD be prepared to ignore the that the RTP header X bit is always zero and SHOULD be
header extension. If a header extension is defined in the prepared to ignore the header extension. If a header
future, that definition MUST specify the contents of the first extension is defined in the future, that definition MUST
16 bits in such a way that multiple different extensions can be specify the contents of the first 16 bits in such a way
identified. that multiple different extensions can be identified.
RTCP packet types: No additional RTCP packet types are defined by RTCP packet types: No additional RTCP packet types are defined
this profile specification. by this profile specification.
RTCP report interval: The suggested constants are to be used for the RTCP report interval: The suggested constants are to be used for
RTCP report interval calculation. Sessions operating under this the RTCP report interval calculation. Sessions operating
profile MAY specify a separate parameter for the RTCP traffic under this profile MAY specify a separate parameter for the
bandwidth rather than using the default fraction of the session RTCP traffic bandwidth rather than using the default
bandwidth. The RTCP traffic bandwidth may be divided into two fraction of the session bandwidth. The RTCP traffic
separate session parameters for those participants which are bandwidth MAY be divided into two separate session
active data senders and those which are not. Following the parameters for those participants which are active data
recommendation in the RTP specification [1] that 1/4 of the RTCP senders and those which are not. Following the
bandwidth be dedicated to data senders, the RECOMMENDED default recommendation in the RTP specification [1] that 1/4 of the
values for these two parameters would be 1.25% and 3.75%, RTCP bandwidth be dedicated to data senders, the
respectively. For a particular session, the RTCP bandwidth for RECOMMENDED default values for these two parameters would
non-data-senders MAY be set to zero when operating on be 1.25% and 3.75%, respectively. For a particular session,
unidirectional links or for sessions that don't require feedback the RTCP bandwidth for non-data-senders MAY be set to zero
on the quality of reception. The RTCP bandwidth for data senders when operating on unidirectional links or for sessions that
SHOULD be kept non-zero so that sender reports can still be sent don't require feedback on the quality of reception. The
for inter-media synchronization and to identify the source by RTCP bandwidth for data senders SHOULD be kept non-zero so
CNAME. The means by which the one or two session parameters for that sender reports can still be sent for inter-media
RTCP bandwidth are specified is beyond the scope of this memo. synchronization and to identify the source by CNAME. The
means by which the one or two session 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 or SR/RR extension: No extension section is defined for the RTCP SR
RR packet. or RR packet.
SDES use: Applications MAY use any of the SDES items described in the SDES use: Applications MAY use any of the SDES items described
RTP specification. While CNAME information MUST be sent every in the RTP specification. While CNAME information MUST be
reporting interval, other items SHOULD only be sent every third sent every reporting interval, other items SHOULD only be
reporting interval, with NAME sent seven out of eight times sent every third reporting interval, with NAME sent seven
within that slot and the remaining SDES items cyclically taking out of eight times within that slot and the remaining SDES
up the eighth slot, as defined in Section 6.2.2 of the RTP items cyclically taking up the eighth slot, as defined in
specification. In other words, NAME is sent in RTCP packets 1, Section 6.2.2 of the RTP specification. In other words,
4, 7, 10, 13, 16, 19, while, say, EMAIL is used in RTCP packet NAME is sent in RTCP packets 1, 4, 7, 10, 13, 16, 19,
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: A user-provided string ("pass phrase") is String-to-key mapping: A user-provided string ("pass phrase") is
hashed with the MD5 algorithm to a 16-octet digest. An n-bit key hashed with the MD5 algorithm to a 16-octet digest. An n-
is extracted from the digest by taking the first n bits from the bit key is extracted from the digest by taking the first n
digest. If several keys are needed with a total length of 128 bits from the digest. If several keys are needed with a
bits or less (as for triple DES), they are extracted in order total length of 128 bits or less (as for triple DES), they
from that digest. The octet ordering is specified in RFC 1423, are extracted in order from that digest. The octet ordering
Section 2.2. (Note that some DES implementations require that is specified in RFC 1423, Section 2.2. (Note that some DES
the 56-bit key be expanded into 8 octets by inserting an odd implementations require that the 56-bit key be expanded
parity bit in the most significant bit of the octet to go with into 8 octets by inserting an odd parity bit in the most
each 7 bits of the key.) significant bit of the octet to go with each 7 bits of the
key.)
It is RECOMMENDED that pass phrases be restricted to ASCII letters, It is RECOMMENDED that pass phrases be restricted to ASCII
digits, the hyphen, and white space to reduce the the chance of letters, digits, the hyphen, and white space to reduce the
transcription errors when conveying keys by phone, fax, telex or the chance of transcription errors when conveying keys by
email. phone, fax, telex or email.
The pass phrase MAY be preceded by a specification of the encryption The pass phrase MAY be preceded by a specification of the
algorithm. Any characters up to the first slash (ASCII 0x2f) are encryption algorithm. Any characters up to the first slash
taken as the name of the encryption algorithm. The encryption format (ASCII 0x2f) are taken as the name of the encryption
specifiers SHOULD be drawn from RFC 1423 or any additional algorithm. The encryption format specifiers SHOULD be drawn
identifiers registered with IANA. If no slash is present, DES-CBC is from RFC 1423 or any additional identifiers registered with
assumed as default. The encryption algorithm specifier is case IANA. If no slash is present, DES-CBC is assumed as
default. The encryption algorithm specifier is case
sensitive. sensitive.
The pass phrase typed by the user is transformed to a canonical form The pass phrase typed by the user is transformed to a
before applying the hash algorithm. For that purpose, we define canonical form before applying the hash algorithm. For that
`white space' to be the ASCII space, formfeed, newline, carriage purpose, we define `white space' to be the ASCII space,
return, tab, or vertical tab as well as all characters contained in formfeed, newline, carriage return, tab, or vertical tab as
the Unicode space characters table. The transformation consists of well as all characters contained in the Unicode space
the following steps: (1) convert the input string to the ISO 10646 characters table. The transformation consists of the
character set, using the UTF-8 encoding as specified in Annex P to following steps: (1) convert the input string to the ISO
ISO/IEC 10646-1:1993 (ASCII characters require no mapping, but ISO 10646 character set, using the UTF-8 encoding as specified
8859-1 characters do); (2) remove leading and trailing white space in Annex P to ISO/IEC 10646-1:1993 (ASCII characters
characters; (3) replace one or more contiguous white space characters require no mapping, but ISO 8859-1 characters do); (2)
by a single space (ASCII or UTF-8 0x20); (4) convert all letters to remove leading and trailing white space characters; (3)
lower case and replace sequences of characters and non-spacing replace one or more contiguous white space characters by a
accents with a single character, where possible. A minimum length of single space (ASCII or UTF-8 0x20); (4) convert all letters
16 key characters (after applying the transformation) SHOULD be to lower case and replace sequences of characters and non-
enforced by the application, while applications MUST allow up to 256 spacing accents with a single character, where possible. A
characters of input. minimum length of 16 key characters (after applying the
transformation) SHOULD be enforced by the application,
while applications MUST allow up to 256 characters of
input.
Underlying protocol: The profile specifies the use of RTP over Underlying protocol: The profile specifies the use of RTP over
unicast and multicast UDP as well as TCP. (This does not unicast and multicast UDP as well as TCP. (This does not
preclude the use of these definitions when RTP is carried by preclude the use of these definitions when RTP is carried
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: A minimal TCP encapsulation is defined. Encapsulation: A minimal TCP encapsulation is defined.
3 Registering Additional Encodings with IANA 3 Registering Additional Encodings with IANA
This profile defines a set of encodings and assigns names to them. This profile lists a set of encodings, each of which is comprised of
These encodings are comprised of a particular media data compression a particular media data compression or representation plus a payload
or representation plus an payload format for encapsulation within format for encapsulation within RTP. Some of those payload formats
RTP. It is expected that additional encodings beyond the set defined are specified here, while others are specified in separate RFCs. It
here will be created in the future. These additional encodings may be is expected that additional encodings beyond the set listed here will
registered with the Internet Assigned Numbers Authority (IANA) as be created in the future and specified in additional payload format
MIME subtype names under the "audio" and "video" MIME types through RFCs.
the MIME registration procedure as specified in RFC 2048 [3].
The MIME registration procedure was originally designed for transport
of multimedia information via asynchronous Internet mail, but the
MIME namespace now provides identification for other transport modes
as well. Some additional parameters are required in the registration
procedure to specify how a particular media type is transported over
RTP. These extensions to the registration procedure, along with
registrations for the encodings defined in this memo, are given in
RFC YYYY [4].
The encoding definitions contained in this memo provide examples of
the information needed:
o the name of the encoding; the names defined here are 3 or 4
characters long to allow a compact representation if needed;
o a description of encoding, including in particular the RTP
timestamp clock rate (or multiple rates for audio encodings
with multiple sampling rates), as may be provided by a separte
RTP payload format specification;
o any additional or optional parameters, such as the number of This profile also assigns to each encoding a short name which MAY be
channels for an audio encoding; used by higher-level control protocols, such as the Session
Description Protocol (SDP), RFC 2327 [5], to identify encodings
selected for a particular RTP session.
o a reference to a further description of the data compression In some contexts it may be useful to refer to these encodings in the
format itself, if available; form of a MIME content-type. To facilitate this, RFC YYYY [3]
provides registrations for all of the encodings names listed here as
MIME subtype names under the "audio" and "video" MIME types through
the MIME registration procedure as specified in RFC 2048 [4].
o contact information and an indication of who has change Any additional encodings specified for use under this profile (or
control over the encoding (this is part of all MIME others) may also be assigned names registered as MIME subtypes with
registrations). the Internet Assigned Numbers Authority (IANA). This registry
provides a means to insure that the names assigned to the additional
encodings are kept unique. RFC YYYY specifies the information that is
required for the registration of RTP encodings.
In addition to assigning names to encodings, this profile also also In addition to assigning names to encodings, this profile also also
assigns static RTP payload types to some of them. However, the assigns static RTP payload type numbers to some of them. However, the
payload type number space is relatively small and cannot accommodate payload type number space is relatively small and cannot accommodate
assignments for all existing and future encodings. During the early assignments for all existing and future encodings. During the early
stages of RTP development, it was necessary to use statically stages of RTP development, it was necessary to use statically
assigned payload types because no other mechanism had been specified assigned payload types because no other mechanism had been specified
to bind encodings to payload types. It was anticipated that non-RTP to bind encodings to payload types. It was anticipated that non-RTP
means beyond the scope of this memo (such as directory services or means beyond the scope of this memo (such as directory services or
invitation protocols) would be specified to establish a dynamic invitation protocols) would be specified to establish a dynamic
mapping between a payload type and an encoding. Now, mechanisms for mapping between a payload type and an encoding. Now, mechanisms for
defining dynamic payload type bindings have been specified in the defining dynamic payload type bindings have been specified in the
Session Description Protocol (SDP), RFC 2327 [5], and in other Session Description Protocol (SDP) and in other protocols such as
protocols such as ITU-T recommendation H.323/H.245. These mechanisms ITU-T recommendation H.323/H.245. These mechanisms associate the
associate the registered name of the encoding/payload format, along registered name of the encoding/payload format, along with any
with any additional required parameters such as the RTP timestamp additional required parameters such as the RTP timestamp clock rate
clock rate and number of channels, to a payload type number. This and number of channels, to a payload type number. This association
association is effective only for the duration of the RTP session in is effective only for the duration of the RTP session in which the
which the dynamic payload type binding is made. This association dynamic payload type binding is made. This association applies only
applies only to the RTP session for which it is made, thus the to the RTP session for which it is made, thus the numbers can be re-
numbers can be re-used for different encodings in different sessions used for different encodings in different sessions so the number
so the number space limitation is avoided. 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. Only applications values in this range for dynamic payload types. Only 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
obtaining session description information that defines the dynamic obtaining session description information that defines the dynamic
payload types. payload types.
Dynamic payload types SHOULD NOT be used without a well-defined Dynamic payload types SHOULD NOT be used without a well-defined
mechanism to indicate the mapping. Systems that expect to mechanism to indicate the mapping. Systems that expect to
interoperate with others operating under this profile SHOULD NOT make interoperate with others operating under this profile SHOULD NOT make
their own assignments of proprietary encodings to particular, fixed their own assignments of proprietary encodings to particular, fixed
payload types. payload types.
skipping to change at page 8, line 35 skipping to change at page 8, line 29
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 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
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 different
channels depends on the encoding. General guidelines are given in channels depends on the encoding. General guidelines are given in
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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 or
for links with severe bandwidth constraints, a higher packetization for links with severe bandwidth constraints, a higher packetization
delay MAY be used. A receiver SHOULD accept packets representing delay MAY be used. A receiver SHOULD accept packets representing
between 0 and 200 ms of audio data. (For framed audio encodings, a between 0 and 200 ms of audio data. (For framed audio encodings, a
receiver SHOULD accept packets with 200 ms divided by the frame receiver SHOULD accept packets with a number of frames equal to 200
duration, rounded up.) This restriction allows reasonable buffer ms divided by the frame duration, rounded up.) This restriction
sizing for the receiver. 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 may
contain any number of audio samples, subject to the constraint that contain any number of audio samples, subject to the constraint that
the number of bits per sample times the number of samples per packet the number of bits per sample times the number of samples per packet
yields an integral octet count. Fractional encodings produce less yields an integral octet count. Fractional encodings produce less
than one octet per sample. than one octet per sample.
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RTP packets SHALL contain a whole number of frames, with frames RTP packets SHALL contain a whole number of frames, with frames
inserted according to age within a packet, so that the oldest frame inserted according to age within a packet, so that the oldest frame
(to be played first) occurs immediately after the RTP packet header. (to be played first) occurs immediately after the RTP packet header.
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 first frame was sampled, that is, the oldest information in the the first frame was sampled, that is, the oldest information in the
packet. packet.
4.5 Audio Encodings 4.5 Audio Encodings
The characteristics of the audio encodings described in this document The characteristics of the audio encodings described in this document
are shown in Table 1; they are listed in order of their payload type
in Table 4. While most audio codecs are only specified for a fixed
sampling rate, some sample-based algorithms (indicated by an entry of
"var." in the sampling rate column of Table 1) may be used with
different sampling rates, resulting in different coded bit rates.
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
name of sampling default name of sampling default
encoding sample/frame bits/sample rate ms/frame ms/packet encoding sample/frame bits/sample rate ms/frame ms/packet
____________________________________________________________________________ __________________________________________________________________
1016 frame N/A 8,000 30 30 1016 frame N/A 8,000 30 30
CN frame N/A var. CN frame N/A var.
DVI4 sample 4 var. 20 DVI4 sample 4 var. 20
G722 sample 8 16,000 20 G722 sample 8 16,000 20
G723 frame N/A 8,000 30 30 G723 frame N/A 8,000 30 30
G726-16 sample 2 8,000 20
G726-24 sample 3 8,000 20
G726-32 sample 4 8,000 20 G726-32 sample 4 8,000 20
G726-40 sample 5 8,000 20
G727-16 sample 2 8,000 20
G727-24 sample 3 8,000 20
G727-32 sample 4 8,000 20
G727-40 sample 5 8,000 20
G728 frame N/A 8,000 2.5 20 G728 frame N/A 8,000 2.5 20
G729 frame N/A 8,000 10 20 G729 frame N/A 8,000 10 20
GSM frame N/A 8,000 20 20 GSM frame N/A 8,000 20 20
GSM-HR frame N/A 8,000 20 20 GSM-HR frame N/A 8,000 20 20
GSM-EFR frame N/A 8,000 20 20 GSM-EFR frame N/A 8,000 20 20
L8 sample 8 var. 20 L8 sample 8 var. 20
L16 sample 16 var. 20 L16 sample 16 var. 20
LPC frame N/A 8,000 20 20 LPC frame N/A 8,000 20 20
MPA frame N/A var. 20 MPA frame N/A var. var.
PCMA sample 8 var. 20 PCMA sample 8 var. 20
PCMU sample 8 var. 20 PCMU sample 8 var. 20
QCELP frame N/A 8,000 20 QCELP frame N/A 8,000 20 20
SX7300P frame N/A 8,000 15 30
SX8300P frame N/A 8,000 15 30
SX9600P frame N/A 8,000 15 30
VDVI sample var. var. 20 VDVI sample var. var. 20
Table 1: Properties of Audio Encodings (N/A: not applicable; var.: Table 1: Properties of Audio Encodings (N/A: not applicable; var.:
variable) variable)
are shown in Table 1; they are listed in order of their payload type
in Table 4. While most audio codecs are only specified for a fixed
sampling rate, some sample-based algorithms (indicated by an entry of
"var." in the sampling rate column of Table 1) may be used with
different sampling rates, resulting in different coded bit rates.
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
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 sampling rate. selected sampling rate.
4.5.1 1016 4.5.1 1016
Encoding 1016 is a frame based encoding using code-excited linear Encoding 1016 is a frame based encoding using code-excited linear
prediction (CELP) and is specified in Federal Standard FED-STD 1016 prediction (CELP) and is specified in Federal Standard FED-STD 1016
[7,8,9,10]. [7,8,9,10].
4.5.2 CN 4.5.2 CN
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Interactive Multimedia Association Interactive Multimedia Association
48 Maryland Avenue, Suite 202 48 Maryland Avenue, Suite 202
Annapolis, MD 21401-8011 Annapolis, MD 21401-8011
USA USA
phone: +1 410 626-1380 phone: +1 410 626-1380
4.5.4 G722 4.5.4 G722
G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding
within 64 kbit/s". within 64 kbit/s". The G.722 encoder produces a stream of octets,
each of which SHALL be octet-aligned in an RTP packet. The first bit
transmitted in the G.722 octet, which is the most significant bit of
the higher sub-band sample, SHALL correspond to the most significant
bit of the octet in the RTP packet.
4.5.5 G723 4.5.5 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 as
a mandatory codec for ITU-T H.324 GSTN videophone terminal 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
skipping to change at page 15, line 5 skipping to change at page 14, line 36
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. This coder was optimized to represent speech the encoder and decoder. This coder was optimized to represent speech
with near-toll quality at the above rates using a limited amount of with near-toll quality at the above rates using a limited amount of
complexity. 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 G.723.1. transmission order of the octets is specified in G.723.1.
4.5.6 G726-16, G726-24, G726-32, G726-40 4.5.6 G726-32
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 32 kbit/s channel. channel encoded at 8000 samples/sec to and from a 32 kbit/s channel.
The conversion is applied to the PCM stream using an Adaptive The conversion is applied to the PCM stream using an Adaptive
Differential Pulse Code Modulation (ADPCM) transcoding technique. Differential Pulse Code Modulation (ADPCM) transcoding technique.
G.726 describes codecs operating at 16 kb/s (2 bits/sample), 24 kb/s G.726 describes codecs operating at 16 kb/s (2 bits/sample), 24 kb/s
(3 bits/sample), 32 kb/s (4 bits/sample), 40 kb/s (5 bits/sample). (3 bits/sample), 32 kb/s (4 bits/sample), 40 kb/s (5 bits/sample).
These encodings are labeled G726-16, G726-24, G726-32 and G726-40, Packetization is specified here only for the 32 kb/s encoding which
respectively. is labeled G726-32.
Note: In 1990, ITU-T Recommendation G.721 was merged with Note: In 1990, ITU-T Recommendation G.721 was merged with
Recommendation G.723 into ITU-T Recommendation G.726. Thus, G726-32 Recommendation G.723 into ITU-T Recommendation G.726. Thus, G726-32
designates the same algorithm as G721 in RFC 1890. designates the same algorithm as G721 in RFC 1890.
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. The 4-bit code words of the G726-32 encoding MUST be the audio data. The 4-bit code words of the G726-32 encoding MUST be
packed into octets as follows: the first code word is placed in the packed into octets as follows: the first code word is placed in the
four least significant bits of the first octet, with the least four least significant bits of the first octet, with the least
significant bit of the code word in the least significant bit of the significant bit of the code word in the least significant bit of the
octet; the second code word is placed in the four most significant octet; the second code word is placed in the four most significant
bits of the first octet, with the most significant bit of the code bits of the first octet, with the most significant bit of the code
word in the most significant bit of the octet. Subsequent pairs of word in the most significant bit of the octet. Subsequent pairs of
the code words SHALL be packed in the same way into successive the code words SHALL be packed in the same way into successive
octets, with the first code word of each pair placed in the least octets, with the first code word of each pair placed in the least
significant four bits of the octet. The number of samples per packet significant four bits of the octet. The number of samples per packet
MUST be even because there is no means to indicate a partially filled MUST be even because there is no means to indicate a partially filled
last octet. last octet.
4.5.7 G727-16, G727-24, G727-32, G727-40 4.5.7 G728
ITU-T Recommendation G.727, "5-, 4-, 3- and 2-bits sample embedded
adaptive differential pulse code modulation (ADPCM)", specifies an
embedded ADPCM algorithm which has the intrinsic capability of
dropping bits in the encoded words to alleviate network congestion
conditions. The algorithm, although not bitstream compatible with
G.726, was based and has a structure similar to the G.726 ADPCM
algorithm.
4.5.8 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 is
called a vector. Four consecutive vectors, labeled V1 to V4 (where V1 called a vector. Four consecutive vectors, labeled V1 to V4 (where V1
is to be played first by the receiver), build one G.728 frame. The is to be played first by the receiver), build one G.728 frame. The
four vectors of 40 bits are packed into 5 octets, labeled B1 through four vectors of 40 bits are packed into 5 octets, labeled B1 through
skipping to change at page 16, line 27 skipping to change at page 16, line 6
<---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
significant bits of V1, the more significant of the two in its MSB, significant bits of V1, the more significant of the two in its MSB,
and the six most significant bits of V2. B1 SHALL be placed first in and the six most significant bits of V2. B1 SHALL be placed first in
the RTP packet and B5 last. the RTP packet and B5 last.
4.5.9 G729 4.5.8 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. The G.729 and G.729 Annex A codecs were
optimized to represent speech with high quality, where G.729 Annex A optimized to represent speech with high quality, where G.729 Annex A
trades some speech quality for an approximate 50% complexity trades some speech quality for an approximate 50% complexity
reduction [12]. reduction [12].
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: while the G.729 Annex B comfort noise frame occupies 2 octets:
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|0 1 2 3 4|0 1 2 3|0 1 2 3 4|S| |F|0 1 2 3 4|0 1 2 3|0 1 2 3 4|S|
skipping to change at page 18, line 5 skipping to change at page 17, line 28
7 7
4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| C2 | S2 | GA2 | GB2 | | C2 | S2 | GA2 | GB2 |
| | | | | | | | | |
|8 9 1 1 1|0 1 2 3|0 1 2|0 1 2 3| |8 9 1 1 1|0 1 2 3|0 1 2|0 1 2 3|
| 0 1 2| | | | | 0 1 2| | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
4.5.10 GSM 4.5.9 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 [13,14,15]. The text of the standard can be obtained rate of 13 kb/s [13,14,15]. 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.
4.5.10.1 General Packaging Issues 4.5.9.1 General Packaging Issues
The GSM standard specifies the bit stream produced by the codec, but The GSM standard (ETS 300 961) specifies the bit stream produced by
does not specify how these bits should be packed for transmission. the codec, but does not specify how these bits should be packed for
Some software implementations of the GSM codec use a different transmission. The packetization specified here has subsequently been
packing than that specified here. adopted in ETSI Technical Specification TS 101 318. Some software
implementations of the GSM codec use a different packing than that
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 4
bit signature (0xD), followed by the MSB encoding of the fields of 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.
4.5.10.2 GSM variable names and numbers 4.5.9.2 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.
4.5.11 GSM-HR 4.5.10 GSM-HR
GSM-HR denotes GSM 06.20 half rate speech transcoding, specified in GSM-HR denotes GSM 06.20 half rate speech transcoding, specified in
ETS 300 969 which is available from ETSI at the address given in ETS 300 969 which is available from ETSI at the address given in
Section 4.5.9. This codec has a frame length of 112 bits (14 octets).
Packing of the fields in the codec bit stream into octets for
transmission in RTP is done in a manner similar to that specified
here for the original GSM 06.10 codec and is specified in ETSI
Technical Specification TS 101 318.
4.5.11 GSM-EFR
GSM-EFR denotes GSM 06.60 enhanced full rate speech transcoding,
specified in ETS 300 969 which is available from ETSI at the address
given in Section 4.5.9. This codec has a frame length of 244 bits.
For transmission in RTP, each codec frame is packed into a 31 octet
(248 bit) buffer beginning with a 4-bit signature 0xC in a manner
similar to that specified here for the original GSM 06.10 codec. The
packing is specified in ETSI Technical Specification TS 101 318.
4.5.12 L8
L8 denotes linear audio data samples, using 8-bits of precision with
an offset of 128, that is, the most negative signal is encoded as
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
9 Nc[0] 7 47 xmc[26] 3 9 Nc[0] 7 47 xmc[26] 3
10 bc[0] 2 48 xmc[27] 3 10 bc[0] 2 48 xmc[27] 3
skipping to change at page 19, line 47 skipping to change at page 19, line 47
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
Section 4.5.10. This codec has a frame length of 112 bits (14 zero.
octets). Packing of the fields in the codec bit stream into octets
for transmission in RTP is done in a manner similar to that specified 4.5.13 L16
here for the original GSM 06.10 codec and is specified in ETSI
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
8 xmc2.1 xmc2.2 xmc3.0 xmc3.1 xmc3.2 xmc4.0 xmc4.1 xmc4.2 8 xmc2.1 xmc2.2 xmc3.0 xmc3.1 xmc3.2 xmc4.0 xmc4.1 xmc4.2
9 xmc5.0 xmc5.1 xmc5.2 xmc6.0 xmc6.1 xmc6.2 xmc7.0 xmc7.1 9 xmc5.0 xmc5.1 xmc5.2 xmc6.0 xmc6.1 xmc6.2 xmc7.0 xmc7.1
skipping to change at page 20, line 42 skipping to change at page 20, 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
Technical Specification TS 101 318.
4.5.12 GSM-EFR
GSM-EFR denotes GSM 06.60 enhanced full rate speech transcoding,
specified in ETS 300 969 which is available from ETSI at the address
given in Section 4.5.10. This codec has a frame length of 244 bits.
For transmission in RTP, each codec frame is packed into a 31 octet
(248 bit) buffer beginning with a 4-bit signature 0xC in a manner
similar to that specified here for the original GSM 06.10 codec. The
packing is specified in ETSI Technical Specification TS 101 318.
4.5.13 L8
L8 denotes linear audio data samples, using 8-bits of precision with
an offset of 128, that is, the most negative signal is encoded as
zero.
4.5.14 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).
4.5.15 LPC 4.5.14 LPC
LPC designates an experimental linear predictive encoding contributed LPC designates an experimental linear predictive encoding contributed
by Ron Frederick, Xerox PARC, which is based on an implementation by Ron Frederick, Xerox PARC, which is based on an implementation
written by Ron Zuckerman, Motorola, posted to the Usenet group written by Ron Zuckerman, Motorola, posted to the Usenet group
comp.dsp on June 26, 1992. The codec generates 14 octets for every comp.dsp on June 26, 1992. The codec generates 14 octets for every
frame. The framesize is set to 20 ms, resulting in a bit rate of frame. The framesize is set to 20 ms, resulting in a bit rate of
5,600 b/s. 5,600 b/s.
4.5.16 MPA 4.5.15 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 [16]. and 13818-3. The encapsulation is specified in RFC 2250 [16].
Sampling rate and channel count are contained in the payload. MPEG-1 The encoding may be at any of three levels of complexity, called
audio supports sampling rates of 32, 44.1, and 48 kHz (ISO/IEC Layer I, II and III. The selected layer as well as the sampling rate
11172-3, section 1.1; "Scope"). MPEG-2 additionally supports sampling and channel count are indicated in the payload. MPEG-1 audio supports
rates of 16, 22.05 and 24 kHz. sampling rates of 32, 44.1, and 48 kHz (ISO/IEC 11172-3, section 1.1;
"Scope"). MPEG-2 supports sampling rates of 16, 22.05 and 24 kHz.
The number of samples per frame is fixed, but the frame size will
vary with the sampling rate and bit rate.
4.5.17 PCMA and PCMU 4.5.16 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 data
is encoded as eight bits per sample, after logarithmic scaling. PCMU is encoded as eight bits per sample, after logarithmic scaling. PCMU
denotes mu-law scaling, PCMA A-law scaling. A detailed description is denotes mu-law scaling, PCMA A-law scaling. A detailed description is
given by Jayant and Noll [17]. Each G.711 octet SHALL be octet- given by Jayant and Noll [17]. Each G.711 octet SHALL be octet-
aligned in an RTP packet. The sign bit of each G.711 octet SHALL aligned in an RTP packet. The sign bit of each G.711 octet SHALL
correspond to the most significant bit of the octet in the RTP packet correspond to the most significant bit of the octet in the RTP packet
(i.e., assuming the G.711 samples are handled as octets on the host (i.e., assuming the G.711 samples are handled as octets on the host
machine, the sign bit SHALL be the most signficant bit of the octet machine, the sign bit SHALL be the most signficant bit of the octet
as defined by the host machine format). The 56 kb/s and 48 kb/s modes as defined by the host machine format). The 56 kb/s and 48 kb/s modes
of G.711 are not applicable to RTP, since PCMA and PCMU SHALL always of G.711 are not applicable to RTP, since PCMA and PCMU SHALL always
be transmitted as 8-bit samples. be transmitted as 8-bit samples.
4.5.18 QCELP 4.5.17 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 of
8000 Hz, 16- bit sampled input speech into one of four different size 8000 Hz, 16- bit sampled input speech into one of four different size
output frames: Rate 1 (266 bits), Rate 1/2 (124 bits), Rate 1/4 (54 output frames: Rate 1 (266 bits), Rate 1/2 (124 bits), Rate 1/4 (54
bits) or Rate 1/8 (20 bits). For typical speech patterns, this 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 [18]. audio codec is described in [18].
4.5.19 RED 4.5.18 RED
The redundant audio payload format "RED" is specified by RFC 2198 The redundant audio payload format "RED" is specified by RFC 2198
[19]. It defines a means by which multiple redundant copies of an [19]. It defines a means by which multiple redundant copies of an
audio packet may be transmitted in a single RTP stream. Each packet audio packet may be transmitted in a single RTP stream. Each packet
in such a stream contains, in addition to the audio data for that in such a stream contains, in addition to the audio data for that
packetization interval, a (more heavily compressed) copy of the data packetization interval, a (more heavily compressed) copy of the data
from a previous packetization interval. This allows an approximation from a previous packetization interval. This allows an approximation
of the data from lost packets to be recovered upon decoding of a of the data from lost packets to be recovered upon decoding of a
subsequent packet, giving much improved sound quality when compared subsequent packet, giving much improved sound quality when compared
with silence substitution for lost packets. with silence substitution for lost packets.
4.5.20 SX* 4.5.19 VDVI
The SX7300P, SX8300P and SX9600P codecs are part of the same
compatible family and distinguished by the first octet in each frame,
where "x" can be any value:
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|0 0 x | SX7300P bitstream (14 byte frame)
|0 1 0 | SX8300P bitstream (16 byte frame)
|1 0 x | VAD bistream ( 2 byte frame)
|1 1 x | SX9600P bitstream (18 byte frame)
+-+-+-+-+-+-+-+-+
4.5.20.1 SX7300P
The SX7300P is a low-complexity CELP-based audio codec operating at a
sampling rate of 8000 Hz. It encodes blocks of 120 audio samples (15
ms) into an encoded frame of 14 octets, yielding an encoded bit rate
of approximately 7467 b/s.
4.5.20.2 SX8300P
The SX8300P is a low-complexity CELP-based audio codec operating at a
sampling rate of 8000 Hz. It encodes blocks of 120 audio samples (15
ms) into an encoded frame of 16 octets, yielding an encoded bit rate
of approximately 8533 b/s.
4.5.20.3 SX9600P
The SX9600P is a low-complexity, toll-quality CELP-based audio codec
operating at a sampling rate of 8000 Hz. It encodes blocks of 120
audio samples (15 ms) into an encoded frame of 18 octets, yielding an
encoded bit rate of 9600 b/s.
4.5.21 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. 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
the valid codewords. The receiver needs to detect the padding
because there is no explicit count of samples in the packet.
It uses the following encoding: It uses the following encoding:
DVI4 codeword VDVI bit pattern DVI4 codeword VDVI bit pattern
_________________________________ _______________________________
0 00 0 00
1 010 1 010
2 1100 2 1100
3 11100 3 11100
4 111100 4 111100
5 1111100 5 1111100
6 11111100 6 11111100
7 11111110 7 11111110
8 10 8 10
9 011 9 011
skipping to change at page 25, line 26 skipping to change at page 24, line 22
packetization and RTP-specific properties are described in RFC 2190 packetization and RTP-specific properties are described in RFC 2190
[24]. [24].
5.6 H263-1998 5.6 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 RFC 2429
[25]. Because the 1998 version of H.263 is a superset of the 1996 [25]. Because the 1998 version of H.263 is a superset of the 1996
syntax, this payload format can also be used with the 1996 version of syntax, this payload format can also be used with the 1996 version of
H.263, and is recommended for this use by new implementations. This H.263, and is RECOMMENDED for this use by new implementations. This
payload format does not replace RFC 2190, which continues to be used payload format does not replace RFC 2190, which continues to be used
by existing implementations, and may be required for backward by existing implementations, and may be required for backward
compatibility in new implementations. Implementations using the new compatibility in new implementations. Implementations using the new
features of the 1998 version of H.263 MUST use the payload format features of the 1998 version of H.263 MUST use the payload format
described in RFC 2429. described in RFC 2429.
5.7 MPV 5.7 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,
skipping to change at page 26, line 49 skipping to change at page 25, line 47
distinguished (see Section "Summary of Protocol Constants" of the RTP distinguished (see Section "Summary of Protocol Constants" of the RTP
protocol specification). 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. payload types within the same media type during a session. See the
section "Multiplexing RTP Sessions" of RFC XXXX for additional
explanation.
Session participants agree through mechanisms beyond the scope of Session participants agree through mechanisms beyond the scope of
this specification on the set of payload types allowed in a given this specification on the set of payload types allowed in a given
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 negotation with a conference control protocol. successful negotation with a conference control protocol.
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 1016 A 8000 1 1 1016 A 8000 1
2 G726-32 A 8000 1 2 G726-32 A 8000 1
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 16000 1 9 G722 A 16000 1
skipping to change at page 28, line 6 skipping to change at page 27, line 6
23 unassigned A 23 unassigned A
dyn GSM-HR A 8000 1 dyn GSM-HR A 8000 1
dyn GSM-EFR A 8000 1 dyn GSM-EFR A 8000 1
dyn RED A dyn RED A
Table 4: Payload types (PT) for audio encodings Table 4: Payload types (PT) for audio encodings
7 RTP over TCP and Similar Byte Stream Protocols 7 RTP over TCP and Similar Byte Stream Protocols
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
skipping to change at page 29, line 25 skipping to change at page 28, line 26
port numbers below 1024 can only be used by privileged processes and port numbers below 1024 can only be used by privileged processes and
port numbers between 1024 and 5000 are automatically assigned by the port numbers between 1024 and 5000 are automatically assigned by the
operating system. operating system.
9 Changes from RFC 1890 9 Changes from RFC 1890
This RFC revises RFC 1890. It is fully backwards-compatible with RFC This RFC revises RFC 1890. It is fully backwards-compatible with RFC
1890 and codifies existing practice. The changes are listed below. 1890 and codifies existing practice. The changes are listed below.
o Additional payload formats and/or expanded descriptions were o Additional payload formats and/or expanded descriptions were
included for CN, G723, G726, G727, G728, G729, GSM, GSM-HR, included for CN, G722, G723, G726, G728, G729, GSM, GSM-HR,
GSM-EFR, QCELP, RED, BT656, H263-1998, MP1S, MP2P and BMPEG. GSM-EFR, QCELP, RED, VDVI, BT656, H263-1998, MP1S, MP2P and
BMPEG.
o Static payload types 4, 12, 16, 17, 18, 19 and 34 were added. o Static payload types 4, 12, 16, 17, 18, 19 and 34 were added.
o The policy is established that no additional registration of o The policy is established that no additional registration of
static payload types for this Profile will be made beyond those static payload types for this Profile will be made beyond
included in Tables 4 and 5, but additional encoding names may those included in Tables 4 and 5, but additional encoding
be registered as MIME subtypes. names may be registered as MIME subtypes.
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". changed from "is" to "SHOULD be".
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.
o This profile follows the suggestion in the RTP spec that RTCP o This profile follows the suggestion in the RTP spec that RTCP
skipping to change at page 30, line 13 skipping to change at page 29, line 15
o A minimal TCP encapsulation is defined. o A minimal TCP encapsulation is defined.
o The security considerations and full copyright sections were o The security considerations and full copyright sections were
added. added.
o According to Peter Hoddie of Apple, only pre-1994 Macintosh o According to Peter Hoddie of Apple, only pre-1994 Macintosh
used the 22254.54 rate and none the 11127.27 rate, so the used the 22254.54 rate and none the 11127.27 rate, so the
latter was dropped from the discussion of suggested sampling latter was dropped from the discussion of suggested sampling
frequencies. frequencies.
o Table 1 was corrected to move some values from the
"ms/packet" column to the "default ms/packet" column where
they belonged.
o Small clarifications of the text have been made in several o Small clarifications of the text have been made in several
places, some in response to questions from readers. In places, some in response to questions from readers. In
particular: particular:
-A definition for "media type" is given in Section 1.1 to allow - A definition for "media type" is given in Section 1.1 to
the explanation of multiplexing RTP sessions in Section 6 to allow the explanation of multiplexing RTP sessions in
be more clear regarding the multiplexing of multiple media. Section 6 to be more clear regarding the multiplexing of
multiple media.
-The explanation of how to determine the number of audio frames - The explanation of how to determine the number of audio
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 items - More description of the allocation of bandwidth to SDES
is given. items is given.
-The terms MUST, SHOULD, MAY, etc. are used as defined in RFC -The terms MUST, SHOULD, MAY, etc. are used as defined in RFC
2119. 2119.
o A second author for this document was added.
10 Security Considerations 10 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 [1]. This profile does not specify any different specification [1]. This profile does not specify any different
security services other than giving rules for mapping characters in a security services other than giving rules for mapping characters in a
user-provided pass phrase to canonical form. The primary function of user-provided pass phrase to canonical form. The primary function of
this profile is to list a set of data compression encodings for audio this profile is to list a set of data compression encodings for audio
and video media. and video media.
skipping to change at page 31, line 5 skipping to change at page 30, line 14
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
computational load. The attacker can inject pathological datagrams computational load. The attacker can inject pathological datagrams
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 [28] and in multicast routing protocols to allow a versions of IGMP [28] 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 11 Full Copyright Statement
Copyright (C) The Internet Society (1999). All Rights Reserved. Copyright (C) The Internet Society (1999). All Rights Reserved.
skipping to change at page 31, line 42 skipping to change at page 30, line 51
This document and the information contained herein is provided on an This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
12 Acknowledgements 12 Acknowledgements
The comments and careful review of Steve Casner, Simao Campos and The comments and careful review of Simao Campos, Richard Cox and AVT
Richard Cox are gratefully acknowledged. The GSM description was Working Group participants are gratefully acknowledged. The GSM
adopted from the IMTC Voice over IP Forum Service Interoperability description was adopted from the IMTC Voice over IP Forum Service
Implementation Agreement (January 1997). Fred Burg and Terry Lyons Interoperability Implementation Agreement (January 1997). Fred Burg
helped with the G.729 description. and Terry Lyons helped with the G.729 description.
13 Address of Author 13 Addresses of Authors
Henning Schulzrinne Henning Schulzrinne
Dept. of Computer Science Dept. of Computer Science
Columbia University Columbia University
1214 Amsterdam Avenue 1214 Amsterdam Avenue
New York, NY 10027 New York, NY 10027
USA USA
electronic mail: schulzrinne@cs.columbia.edu electronic mail: schulzrinne@cs.columbia.edu
Stephen L. Casner
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134
United States
electronic mail: casner@cisco.com
A Bibliography A Bibliography
[1] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A [1] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A
transport protocol for real-time applications," Internet Draft, transport protocol for real-time applications," Internet Draft,
Internet Engineering Task Force, Feb. 1999 Work in progress, revision Internet Engineering Task Force, Feb. 1999 Work in progress, revision
to RFC 1889. to RFC 1889.
[2] S. Bradner, "Key words for use in RFCs to Indicate Requirement [2] S. Bradner, "Key words for use in RFCs to Indicate Requirement
Levels," RFC 2119, Internet Engineering Task Force, Mar. 1997. Levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.
[3] N. Freed, J. Klensin, and J. Postel, "Multipurpose Internet Mail [3] P. Hoschka, "MIME Type Registration of RTP Payload Types,"
Extensions (MIME) Part Four: Registration Procedures," RFC 2048,
Internet Engineering Task Force, Nov. 1996.
[4] P. Hoschka, "MIME Type Registration of RTP Payload Types,"
Internet Draft, Internet Engineering Task Force, Feb. 1999 Work in Internet Draft, Internet Engineering Task Force, Feb. 1999 Work in
progress. progress.
[4] N. Freed, J. Klensin, and J. Postel, "Multipurpose Internet Mail
Extensions (MIME) Part Four: Registration Procedures," RFC 2048,
Internet Engineering Task Force, Nov. 1996.
[5] M. Handley and V. Jacobson, "SDP: Session Description Protocol," [5] M. Handley and V. Jacobson, "SDP: Session Description Protocol,"
Request for Comments (Proposed Standard) RFC 2327, Internet Request for Comments (Proposed Standard) RFC 2327, Internet
Engineering Task Force, Apr. 1998. Engineering Task Force, Apr. 1998.
[6] Apple Computer, "Audio interchange file format AIFF-C," Aug. [6] Apple Computer, "Audio interchange file format AIFF-C," Aug.
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).
[7] Office of Technology and Standards, "Telecommunications: Analog [7] Office of Technology and Standards, "Telecommunications: Analog
to digital conversion of radio voice by 4,800 bit/second code excited to digital conversion of radio voice by 4,800 bit/second code excited
linear prediction (celp)," Federal Standard FS-1016, GSA, Room 6654; linear prediction (celp)," Federal Standard FS-1016, GSA, Room 6654;
skipping to change at page 35, line 32 skipping to change at page 35, line 5
G723 G723
The reference C code implementation defining the G.723.1 algorithm The reference C code implementation defining the G.723.1 algorithm
and its Annexes A, B, and C are available as an integral part of and its Annexes A, B, and C are available as an integral part of
Recommendation G.723.1 from the ITU Sales Service, address listed Recommendation G.723.1 from the ITU Sales Service, address listed
above. Both the algorithm and C code are covered by a specific above. Both the algorithm and C code are covered by a specific
license. The ITU-T Secretariat should be contacted to obtain such license. The ITU-T Secretariat should be contacted to obtain such
licensing information. licensing information.
G726-16 through G726-40 G726-32
G726-16 through G726-40 are specified in the ITU-T Recommendation
G.726, "40, 32, 24, and 16 kb/s Adaptive Differential Pulse Code
Modulation (ADPCM)". An implementation of the G.726 algorithm is
available as part of the ITU-T STL, described above.
G727-16 through G727-40
G727-16 through G727-40 are specified in the ITU-T Recommendation G726-32 is specified in the ITU-T Recommendation G.726, "40, 32, 24,
G.727, "5-, 4-, 3-, and 2-bit/sample embedded adaptive differential and 16 kb/s Adaptive Differential Pulse Code Modulation (ADPCM)". An
pulse code modulation". An implementation of the G.727 algorithm will implementation of the G.726 algorithm is available as part of the
be available in a future release of the ITU-T STL, described above. ITU-T STL, described above.
G729 G729
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 and B are available as an integral part of its Annexes A and B are available as an integral part of
Recommendation G.729 from the ITU Sales Service, listed above. Both Recommendation G.729 from the ITU Sales Service, listed above. Both
the algorithm and the C code are covered by a specific license. The the algorithm and the C code are covered by a specific license. The
contact information for obtaining the license is listed in the C contact information for obtaining the license is listed in the C
code. code.
skipping to change at page 37, line 9 skipping to change at page 36, line 14
1 Introduction ........................................ 2 1 Introduction ........................................ 2
1.1 Terminology ......................................... 3 1.1 Terminology ......................................... 3
2 RTP and RTCP Packet Forms and Protocol Behavior ..... 3 2 RTP and RTCP Packet Forms and Protocol Behavior ..... 3
3 Registering Additional Encodings with IANA .......... 6 3 Registering Additional Encodings with IANA .......... 6
4 Audio ............................................... 7 4 Audio ............................................... 7
4.1 Encoding-Independent Rules .......................... 7 4.1 Encoding-Independent Rules .......................... 7
4.2 Operating Recommendations ........................... 9 4.2 Operating Recommendations ........................... 9
4.3 Guidelines for Sample-Based Audio Encodings ......... 9 4.3 Guidelines for Sample-Based Audio Encodings ......... 9
4.4 Guidelines for Frame-Based Audio Encodings .......... 10 4.4 Guidelines for Frame-Based Audio Encodings .......... 10
4.5 Audio Encodings ..................................... 10 4.5 Audio Encodings ..................................... 10
4.5.1 1016 ................................................ 12 4.5.1 1016 ................................................ 11
4.5.2 CN .................................................. 12 4.5.2 CN .................................................. 11
4.5.3 DVI4 ................................................ 13 4.5.3 DVI4 ................................................ 12
4.5.4 G722 ................................................ 14 4.5.4 G722 ................................................ 13
4.5.5 G723 ................................................ 14 4.5.5 G723 ................................................ 13
4.5.6 G726-16, G726-24, G726-32, G726-40 .................. 15 4.5.6 G726-32 ............................................. 14
4.5.7 G727-16, G727-24, G727-32, G727-40 .................. 15 4.5.7 G728 ................................................ 15
4.5.8 G728 ................................................ 15 4.5.8 G729 ................................................ 16
4.5.9 G729 ................................................ 16 4.5.9 GSM ................................................. 17
4.5.10 GSM ................................................. 18 4.5.9.1 General Packaging Issues ............................ 17
4.5.10.1 General Packaging Issues ............................ 18 4.5.9.2 GSM variable names and numbers ...................... 18
4.5.10.2 GSM variable names and numbers ...................... 18 4.5.10 GSM-HR .............................................. 18
4.5.11 GSM-HR .............................................. 18 4.5.11 GSM-EFR ............................................. 18
4.5.12 GSM-EFR ............................................. 20 4.5.12 L8 .................................................. 18
4.5.13 L8 .................................................. 21 4.5.13 L16 ................................................. 19
4.5.14 L16 ................................................. 21 4.5.14 LPC ................................................. 20
4.5.15 LPC ................................................. 21 4.5.15 MPA ................................................. 21
4.5.16 MPA ................................................. 21 4.5.16 PCMA and PCMU ....................................... 21
4.5.17 PCMA and PCMU ....................................... 21 4.5.17 QCELP ............................................... 21
4.5.18 QCELP ............................................... 22 4.5.18 RED ................................................. 22
4.5.19 RED ................................................. 22 4.5.19 VDVI ................................................ 22
4.5.20 SX* ................................................. 22 5 Video ............................................... 22
4.5.20.1 SX7300P ............................................. 22 5.1 BT656 ............................................... 23
4.5.20.2 SX8300P ............................................. 23 5.2 CelB ................................................ 23
4.5.20.3 SX9600P ............................................. 23 5.3 JPEG ................................................ 23
4.5.21 VDVI ................................................ 23 5.4 H261 ................................................ 23
5 Video ............................................... 24 5.5 H263 ................................................ 24
5.1 BT656 ............................................... 24 5.6 H263-1998 ........................................... 24
5.2 CelB ................................................ 24 5.7 MPV ................................................. 24
5.3 JPEG ................................................ 24 5.8 MP2T ................................................ 24
5.4 H261 ................................................ 25 5.9 MP1S ................................................ 24
5.5 H263 ................................................ 25 5.10 MP2P ................................................ 24
5.6 H263-1998 ........................................... 25 5.11 BMPEG ............................................... 25
5.7 MPV ................................................. 25 5.12 nv .................................................. 25
5.8 MP2T ................................................ 25 6 Payload Type Definitions ............................ 25
5.9 MP1S ................................................ 25 7 RTP over TCP and Similar Byte Stream Protocols ...... 26
5.10 MP2P ................................................ 25 8 Port Assignment ..................................... 27
5.11 BMPEG ............................................... 26 9 Changes from RFC 1890 ............................... 28
5.12 nv .................................................. 26 10 Security Considerations ............................. 29
6 Payload Type Definitions ............................ 26 11 Full Copyright Statement ............................ 30
7 RTP over TCP and Similar Byte Stream Protocols ...... 27 12 Acknowledgements .................................... 30
8 Port Assignment ..................................... 28 13 Addresses of Authors ................................ 31
9 Changes from RFC 1890 ............................... 29 A Bibliography ........................................ 31
10 Security Considerations ............................. 30
11 Full Copyright Statement ............................ 31
12 Acknowledgements .................................... 31
13 Address of Author ................................... 31
A Bibliography ........................................ 32
 End of changes. 

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