RFC 4867
RTP Payload Format and File Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs
6. Congestion Control
The general congestion control considerations for transporting RTP data apply to AMR or AMR-WB speech over RTP as well. However, the multi-rate capability of AMR and AMR-WB speech coding may provide an advantage over other payload formats for controlling congestion since the bandwidth demand can be adjusted by selecting a different coding mode. Another parameter that may impact the bandwidth demand for AMR and AMR-WB is the number of frame-blocks that are encapsulated in each RTP payload. Packing more frame-blocks in each RTP payload can reduce the number of packets sent and hence the overhead from IP/UDP/RTP headers, at the expense of increased delay. If forward error correction (FEC) is used to combat packet loss, the amount of redundancy added by FEC will need to be regulated so that the use of FEC itself does not cause a congestion problem. It is RECOMMENDED that AMR or AMR-WB applications using this payload format employ congestion control. The actual mechanism for congestion control is not specified but should be suitable for real- time flows, possibly "TCP Friendly Rate Control" [21].7. Security Considerations
RTP packets using the payload format defined in this specification are subject to the general security considerations discussed in [8] and in any used profile, like AVP [12] or SAVP [26]. As this format transports encoded speech, the main security issues include confidentiality, authentication, and integrity of the speech itself. The payload format itself does not have any built-in security mechanisms. External mechanisms, such as SRTP [26], need to be used for this functionality. Note that the appropriate mechanism to provide security to RTP and the payloads following this memo may vary. It is dependent on the application, the transport, and the signaling protocol employed. Therefore, a single mechanism is not sufficient, although if suitable the usage of SRTP [26] is RECOMMENDED. Other known mechanisms that may be used are IPsec [33] and TLS [34] (RTP over TCP), but other alternatives may also exist. This payload format does not exhibit any significant non-uniformity in the receiver side computational complexity for packet processing, and thus is unlikely to pose a denial-of-service threat due to the receipt of pathological data.
7.1. Confidentiality
To achieve confidentiality of the encoded AMR or AMR-WB speech, all speech data bits will need to be encrypted. There is less of a need to encrypt the payload header or the table of contents due to a) that they only carry information about the requested speech mode, frame type, and frame quality, and b) that this information could be useful to some third party, e.g., quality monitoring. The packetization and unpacketization of the AMR and AMR-WB payload is done only at the endpoints. Therefore encryption should be performed after packet encapsulation, and decryption should be performed before packet decapsulation. Encryption may affect interleaving. Specifically, a change of keys should occur at the boundary between interleaving groups. If it is not done at that boundary on both endpoints, the speech quality will be degraded during the complete interleaving group for any receiver. The encryption mechanism may impact the robustness of the error correcting mechanism. This is discussed in Section 9.5 of SRTP [26]. From this, UED/UEP based on robust sorting may be difficult to apply when the payload data is encrypted.7.2. Authentication and Integrity
To authenticate the sender and to protect the integrity of the RTP packets in transit, an external mechanism has to be used. As stated before, it is RECOMMENDED that SRTP [26] be used for common interoperability. Note that the use of UED/UEP may be difficult to combine with some integrity protection mechanisms because any bit errors will cause the integrity check to fail. Data tampering by a man-in-the-middle attacker could result in erroneous depacketization/decoding that could lower the speech quality or produce unintelligible communications. Tampering with the CMR field may result in a different speech quality than desired.8. Payload Format Parameters
This section defines the parameters that may be used to select optional features of the AMR and AMR-WB payload formats. The parameters are defined here as part of the media type registrations for the AMR and AMR-WB speech codecs. The registrations are done following RFC 4855 [15] and the media registration rules [14].
A mapping of the parameters into the Session Description Protocol (SDP) [11] is also provided for those applications that use SDP. Equivalent parameters could be defined elsewhere for use with control protocols that do not use media types or SDP. Two separate media type registrations are made, one for AMR and one for AMR-WB, because they are distinct encodings that must be distinguished by their own media type. Data formats are specified for both real-time transport in RTP and for storage type applications such as email attachments.8.1. AMR Media Type Registration
The media type for the Adaptive Multi-Rate (AMR) codec is allocated from the IETF tree since AMR is a widely used speech codec in general VoIP and messaging applications. This media type registration covers both real-time transfer via RTP and non-real-time transfers via stored files. Note, any unspecified parameter MUST be ignored by the receiver. Media Type name: audio Media subtype name: AMR Required parameters: none Optional parameters: These parameters apply to RTP transfer only. octet-align: Permissible values are 0 and 1. If 1, octet-aligned operation SHALL be used. If 0 or if not present, bandwidth-efficient operation is employed. mode-set: Restricts the active codec mode set to a subset of all modes, for example, to be able to support transport channels such as GSM networks in gateway use cases. Possible values are a comma separated list of modes from the set: 0,...,7 (see Table 1a [2]). The SID frame type 8 and NO_DATA (frame type 15) are never included in the mode set, but can always be used. If mode-set is specified, it MUST be abided, and frames encoded with modes outside of the subset MUST NOT be sent in any RTP payload or used in codec mode requests. If not present, all codec modes are allowed for the payload type.
mode-change-period: Specifies a number of frame-blocks, N (1 or
2), that is the frame-block period at which codec mode
changes are allowed for the sender. The initial phase of
the interval is arbitrary, but changes must be separated
by a period of N frame-blocks, i.e., a value of 2
allows the sender to change mode every second frame-
block. The value of N SHALL be either 1 or 2. If this
parameter is not present, mode changes are allowed at
any time during the session, i.e., N=1.
mode-change-capability: Specifies if the client is capable to
transmit with a restricted mode change period. The
parameter may take value of 1 or 2. A value of 1
indicates that the client is not capable of restricting
the mode change period to 2, and that the codec mode may
be changed at any point. A value of 2 indicates that the
client has the capability to restrict the mode change
period to 2, and thus that the client can correctly
interoperate with a receiver requiring a mode-change-
period=2. If this parameter is not present, the mode-
change restriction capability is not supported, i.e.
mode-change-capability=1. To be able to interoperate
fully with gateways to circuit switched networks (for
example, GSM networks), transmissions with restricted
mode changes (mode-change-capability=2) are required.
Thus, clients RECOMMENDED to have the capability to
support transmission according to
mode-change-capability=2.
mode-change-neighbor: Permissible values are 0 and 1. If 1, the
sender SHOULD only perform mode changes to the
neighboring modes in the active codec mode set.
Neighboring modes are the ones closest in bit rate to
the current mode, either the next higher or next lower
rate. If 0 or if not present, change between any two
modes in the active codec mode set is allowed.
maxptime: The maximum amount of media which can be encapsulated
in a payload packet, expressed as time in milliseconds.
The time is calculated as the sum of the time that the
media present in the packet represents. The time SHOULD
be an integer multiple of the frame size. If this
parameter is not present, the sender MAY encapsulate any
number of speech frames into one RTP packet.
crc: Permissible values are 0 and 1. If 1, frame CRCs SHALL be
included in the payload. If 0 or not present, CRCs
SHALL NOT be used. If crc=1, this also implies
automatically that octet-aligned operation SHALL be used
for the session.
robust-sorting: Permissible values are 0 and 1. If 1, the
payload SHALL employ robust payload sorting. If 0 or if
not present, simple payload sorting SHALL be used. If
robust-sorting=1, this also implies automatically that
octet-aligned operation SHALL be used for the session.
interleaving: Indicates that frame-block level interleaving SHALL
be used for the session, and its value defines the
maximum number of frame-blocks allowed in an
interleaving group (see Section 4.4.1). If this
parameter is not present, interleaving SHALL NOT be
used. The presence of this parameter also implies
automatically that octet-aligned operation SHALL be
used.
ptime: see RFC 4566 [11].
channels: The number of audio channels. The possible values
(1-6) and their respective channel order is specified in
Section 4.1 in [12]. If omitted, it has the default
value of 1.
max-red: The maximum duration in milliseconds that elapses between
the primary (first) transmission of a frame and any
redundant transmission that the sender will use. This
parameter allows a receiver to have a bounded delay when
redundancy is used. Allowed values are between 0 (no
redundancy will be used) and 65535. If the parameter is
omitted, no limitation on the use of redundancy is
present.
Encoding considerations:
The Audio data is binary data, and must be encoded for non-
binary transport; the Base64 encoding is suitable for email.
When used in RTP context the data is framed as defined in [14].
Security considerations:
See Section 7 of RFC 4867.
Public specification:
RFC 4867
3GPP TS 26.090, 26.092, 26.093, 26.101
Applications that use this media type:
This media type is used in numerous applications needing
transport or storage of encoded voice. Some examples include;
Voice over IP, streaming media, voice messaging, and voice
recording on digital cameras.
Additional information:
The following applies to stored-file transfer methods:
Magic numbers:
single-channel:
ASCII character string "#!AMR\n"
(or 0x2321414d520a in hexadecimal)
multi-channel:
ASCII character string "#!AMR_MC1.0\n"
(or 0x2321414d525F4D43312E300a in hexadecimal)
File extensions: amr, AMR
Macintosh file type code: "amr " (fourth character is space)
AMR speech frames may also be stored in the file format "3GP"
defined in 3GPP TS 26.244 [31], which is identified using the
media types "audio/3GPP" or "video/3GPP" as registered by RFC
3839 [32].
Person & email address to contact for further information:
Magnus Westerlund <magnus.westerlund@ericsson.com>
Ari Lakaniemi <ari.lakaniemi@nokia.com>
Intended usage: COMMON.
This media type is widely used in streaming, VoIP, and messaging
applications on many types of devices.
Restrictions on usage:
When this media type is used in the context of transfer over
RTP, the RTP payload format specified in Section 4 SHALL be
used. In all other contexts, the file format defined in Section
5 SHALL be used.
Author:
Magnus Westerlund <magnus.westerlund@ericsson.com>
Ari Lakaniemi <ari.lakaniemi@nokia.com>
Change controller:
IETF Audio/Video Transport working group delegated from the
IESG.
8.2. AMR-WB Media Type Registration
The media type for the Adaptive Multi-Rate Wideband (AMR-WB) codec is allocated from the IETF tree since AMR-WB is a widely used speech codec in general VoIP and messaging applications. This media type registration covers both real-time transfer via RTP and non-real- time transfers via stored files. Note, any unspecified parameter MUST be ignored by the receiver. Media Type name: audio Media subtype name: AMR-WB Required parameters: none Optional parameters: These parameters apply to RTP transfer only. octet-align: Permissible values are 0 and 1. If 1, octet-aligned operation SHALL be used. If 0 or if not present, bandwidth-efficient operation is employed. mode-set: Restricts the active codec mode set to a subset of all modes, for example, to be able to support transport channels such as GSM networks in gateway use cases. Possible values are a comma-separated list of modes from the set: 0,...,8 (see Table 1a [4]). The SID frame type 9, SPEECH_LOST (frame type 14), and NO_DATA (frame type 15) are never included in the mode set, but can always be used. If mode-set is specified, it MUST be abided, and frames encoded with modes outside of the subset MUST NOT be sent in any RTP payload or used in codec mode requests. If not present, all codec modes are allowed for the payload type. mode-change-period: Specifies a number of frame-blocks, N (1 or 2), that is the frame-block period at which codec mode changes are allowed for the sender. The initial phase of the interval is arbitrary, but changes must be separated by multiples of N frame-blocks, i.e., a value of 2 allows the sender to change mode every second frame- block. The value of N SHALL be either 1 or 2. If this parameter is not present, mode changes are allowed at Any time during the session, i.e., N=1.
mode-change-capability: Specifies if the client is capable to
transmit with a restricted mode change period. The
parameter may take value of 1 or 2. A value of 1
indicates that the client is not capable of restricting
the mode change period to 2, and that the codec mode may
be changed at any point. A value of 2 indicates that the
client has the capability to restrict the mode change
period to 2, and thus that the client can correctly
interoperate with a receiver requiring a mode-change-
period=2. If this parameter is not present, the mode-
change restriction capability is not supported, i.e.
mode-change-capability=1. To be able to interoperate
fully with gateways to circuit switched networks (for
example, GSM networks), transmissions with restricted
mode changes (mode-change-capability=2) are required.
Thus, clients are RECOMMENDED to have the capability to
support transmission according to
mode-change-capability=2.
mode-change-neighbor: Permissible values are 0 and 1. If 1, the
sender SHOULD only perform mode changes to the
neighboring modes in the active codec mode set.
Neighboring modes are the ones closest in bit rate to
the current mode, either the next higher or next lower
rate. If 0 or if not present, change between any two
modes in the active codec mode set is allowed.
maxptime: The maximum amount of media which can be encapsulated
in a payload packet, expressed as time in milliseconds.
The time is calculated as the sum of the time that the
media present in the packet represents. The time SHOULD
be an integer multiple of the frame size. If this
parameter is not present, the sender MAY encapsulate any
number of speech frames into one RTP packet.
crc: Permissible values are 0 and 1. If 1, frame CRCs SHALL be
included in the payload. If 0 or not present, CRCs
SHALL NOT be used. If crc=1, this also implies
automatically that octet-aligned operation SHALL be used
for the session.
robust-sorting: Permissible values are 0 and 1. If 1, the
payload SHALL employ robust payload sorting. If 0 or if
not present, simple payload sorting SHALL be used. If
robust-sorting=1, this also implies automatically that
octet-aligned operation SHALL be used for the session.
interleaving: Indicates that frame-block level interleaving SHALL
be used for the session, and its value defines the
maximum number of frame-blocks allowed in an
interleaving group (see Section 4.4.1). If this
parameter is not present, interleaving SHALL NOT be
used. The presence of this parameter also implies
automatically that octet-aligned operation SHALL be
used.
ptime: see RFC 2327 [11].
channels: The number of audio channels. The possible values
(1-6) and their respective channel order is specified in
Section 4.1 in [12]. If omitted, it has the default
value of 1.
max-red: The maximum duration in milliseconds that elapses between
the primary (first) transmission of a frame and any
redundant transmission that the sender will use. This
parameter allows a receiver to have a bounded delay when
redundancy is used. Allowed values are between 0 (no
redundancy will be used) and 65535. If the parameter is
omitted, no limitation on the use of redundancy is
present.
Encoding considerations:
The Audio data is binary data, and must be encoded for non-
binary transport; the Base64 encoding is suitable for email.
When used in RTP context the data is framed as defined in [14].
Security considerations:
See Section 7 of RFC 4867.
Public specification:
RFC 4867
3GPP TS 26.190, 26.192, 26.193, 26.201
Applications that use this media type:
This media type is used in numerous applications needing
transport or storage of encoded voice. Some examples include;
Voice over IP, streaming media, voice messaging, and voice
recording on digital cameras.
Additional information:
The following applies to stored-file transfer methods:
Magic numbers:
single-channel:
ASCII character string "#!AMR-WB\n"
(or 0x2321414d522d57420a in hexadecimal)
multi-channel:
ASCII character string "#!AMR-WB_MC1.0\n"
(or 0x2321414d522d57425F4D43312E300a in hexadecimal)
File extensions: awb, AWB
Macintosh file type code: amrw
Object identifier or OID: none
AMR-WB speech frames may also be stored in the file format "3GP"
defined in 3GPP TS 26.244 [31] and identified using the media
type "audio/3GPP" or "video/3GPP" as registered by RFC 3839
[32].
Person & email address to contact for further information:
Magnus Westerlund <magnus.westerlund@ericsson.com>
Ari Lakaniemi <ari.lakaniemi@nokia.com>
Intended usage: COMMON.
This media type is widely used in streaming, VoIP, and messaging
applications on many types of devices.
Restrictions on usage:
When this media type is used in the context of transfer over
RTP, the RTP payload format specified in Section 4 SHALL be
used. In all other contexts, the file format defined in Section
5 SHALL be used.
Author:
Magnus Westerlund <magnus.westerlund@ericsson.com>
Ari Lakaniemi <ari.lakaniemi@nokia.com>
Change controller:
IETF Audio/Video Transport working group delegated from the
IESG.
8.3. Mapping Media Type Parameters into SDP
The information carried in the media type specification has a
specific mapping to fields in the Session Description Protocol (SDP)
[11], which is commonly used to describe RTP sessions. When SDP is
used to specify sessions employing the AMR or AMR-WB codec, the
mapping is as follows:
- The media type ("audio") goes in SDP "m=" as the media name.
- The media subtype (payload format name) goes in SDP "a=rtpmap"
as the encoding name. The RTP clock rate in "a=rtpmap" MUST be
8000 for AMR and 16000 for AMR-WB, and the encoding parameters
(number of channels) MUST either be explicitly set to N or
omitted, implying a default value of 1. The values of N that
are allowed are specified in Section 4.1 in [12].
- The parameters "ptime" and "maxptime" go in the SDP "a=ptime"
and "a=maxptime" attributes, respectively.
- Any remaining parameters go in the SDP "a=fmtp" attribute by
copying them directly from the media type parameter string as a
semicolon-separated list of parameter=value pairs.
8.3.1. Offer-Answer Model Considerations
The following considerations apply when using SDP Offer-Answer
procedures to negotiate the use of AMR or AMR-WB payload in RTP:
- Each combination of the RTP payload transport format
configuration parameters (octet-align, crc, robust-sorting,
interleaving, and channels) is unique in its bit-pattern and
not compatible with any other combination. When creating an
offer in an application desiring to use the more advanced
features (crc, robust-sorting, interleaving, or more than one
channel), the offerer is RECOMMENDED to also offer a payload
type containing only the octet-aligned or bandwidth-efficient
configuration with a single channel. If multiple
configurations are of interest to the application, they may all
be offered; however, care should be taken not to offer too many
payload types. An SDP answerer MUST include, in the SDP answer
for a payload type, the following parameters unmodified from
the SDP offer (unless it removes the payload type): "octet-
align"; "crc"; "robust-sorting"; "interleaving"; and
"channels". The SDP offerer and answerer MUST generate AMR or
AMR-WB packets as described by these parameters.
- The "mode-set" parameter can be used to restrict the set of
active AMR/AMR-WB modes used in a session. This functionality
is primarily intended for gateways to access networks such as
GSM or 3GPP UMTS, where the access network may be capable of
supporting only a subset of AMR/AMR-WB modes. The 3GPP
preferred codec configurations are defined in 3GPP TS 26.103
[25], and it is RECOMMENDED that other networks also needing to
restrict the mode set follow the preferred codec configurations
defined in 3GPP for greatest interoperability.
The parameter is bi-directional, i.e., the restricted set
applies to media both to be received and sent by the declaring
entity. If a mode set was supplied in the offer, the answerer
SHALL return the mode-set unmodified or reject the payload
type. However, the answerer is free to choose a mode-set in
the answer only if no mode-set was supplied in the offer for a
unicast two-peer session. The mode-set in the answer is
binding both for offerer and answerer. Thus, an offerer
supporting all modes and subsets SHOULD NOT include the mode-
set parameter. For any other offerer it is RECOMMENDED to
include each mode-set it can support as a separate payload type
within the offer. For multicast sessions, the answerer SHALL
only participate in the session if it supports the offered
mode-set. Thus, it is RECOMMENDED that any offer for a
multicast session include only the mode-set it will require the
answerers to support, and that the mode-set be likely to be
supported by all participants.
- The parameters "mode-change-period" and "mode-change-
capability" are intended to be used in sessions with gateways,
for example, when interoperating with GSM networks. Both
parameters are declarative and are combined to allow a session
participant to determine if the payload type can be supported.
The mode-change-period will indicate what the offerer or
answerer requires of data it receives, while the mode-change-
capability indicates its transmission capabilities.
A mode-change-period=2 in the offer indicates a requirement on
the answerer to send with a mode-change period of 2, i.e.,
support mode-change-capability=2. If the answerer requires
mode-change-period=2, it SHALL only include it in the answer if
the offerer either has indicated support with mode-change-
capability=2 or has indicated mode-change-period=2; otherwise,
the payload type SHALL be rejected. An offerer that supports
mode-change-capability=2 SHALL include the parameter in all
offers to ensure the greatest possible interoperability, unless
it includes mode-change-period=2 in the offer. The mode-
change-capability SHOULD be included in answers. It is then
indicating the answerer's capability to transmit with that
mode-change-period for the provided payload format
configuration. The information is useful in future
re-negotiation of the payload formats.
- The parameter "mode-change-neighbor" is a recommendation to
restrict the switching of codec modes to its neighbor and
SHOULD be followed. It is intended to be used in gateway
scenarios (for example, to GSM networks) where the support of
this parameter and the operations it implies improves
interoperability.
"mode-change-neighbor" is a declarative parameter. By
including the parameter, the offerer or answerer indicates that
it desires to receive streams with "mode-change-neighbor"
restrictions.
- In most cases, the parameters "maxptime" and "ptime" will not
affect interoperability; however, the setting of the parameters
can affect the performance of the application. The SDP offer-
answer handling of the "ptime" parameter is described in RFC
3264 [13]. The "maxptime" parameter MUST be handled in the
same way.
- The parameter "max-red" is a stream property parameter. For
send-only or send-recv unicast media streams, the parameter
declares the limitation on redundancy that the stream sender
will use. For recvonly streams, it indicates the desired value
for the stream sent to the receiver. The answerer MAY change
the value, but is RECOMMENDED to use the same limitation as the
offer declares. In the case of multicast, the offerer MAY
declare a limitation; this SHALL be answered using the same
value. A media sender using this payload format is RECOMMENDED
to always include the "max-red" parameter. This information is
likely to simplify the media stream handling in the receiver.
This is especially true if no redundancy will be used, in which
case "max-red" is set to 0. As this parameter was not defined
originally, some senders will not declare this parameter even
if it will limit or not send redundancy at all.
- Any unknown parameter in an offer SHALL be removed in the
answer.
8.3.2. Usage of Declarative SDP
In declarative usage, like SDP in RTSP [29] or SAP [30], the
parameters SHALL be interpreted as follows:
- The payload format configuration parameters (octet-align, crc,
robust-sorting, interleaving, and channels) are all declarative,
and a participant MUST use the configuration(s) that is provided
for the session. More than one configuration may be provided if
necessary by declaring multiple RTP payload types; however, the
number of types should be kept small.
- Any restriction of the AMR or AMR-WB encoder mode-switching and
mode usage through the "mode-set", and "mode-change-period" MUST
be followed by all participants of the session. The restriction
indicated by "mode-change-neighbor" SHOULD be followed. Please
note that such restrictions may be necessary if gateways to other
transport systems like GSM participate in the session. Failure to
consider such restrictions may result in failure for a peer behind
such a gateway to correctly receive all or parts of the session.
Also, if different restrictions are needed by different peers in
the same session (unless a common subset of the restrictions
exists), some peer will not be able to participate. Note that the
usage of mode-change-capability is meaningless when no negotiation
exists, and can thus be excluded in any declarations.
- Any "maxptime" and "ptime" values should be selected with care to
ensure that the session's participants can achieve reasonable
performance.
- The usage of "max-red" puts a global upper limit on the usage of
redundancy that needs to be followed by all that understand the
parameter. However, due to the late addition of this parameter,
it may be ignored by some implementations.
8.3.3. Examples
Some example SDP session descriptions utilizing AMR and AMR-WB
encodings follow. In these examples, long a=fmtp lines are folded to
meet the column width constraints of this document; the backslash
("\") at the end of a line and the carriage return that follows it
should be ignored.
In an example of the usage of AMR in a possible GSM gateway-to-
gateway scenario, the offerer is capable of supporting three
different mode-sets and needs the mode-change-period to be 2 in
combination with mode-change-neighbor restrictions. The other
gateway can only support two of these mode-sets and removes the
payload type 97 in the answer. If the offering GSM gateway only
supports a single mode-set active at the same time, it should
consider doing the 1 out of N selection procedures described in
Section 10.2 of [13]:
Offer:
m=audio 49120 RTP/AVP 97 98 99
a=rtpmap:97 AMR/8000/1
a=fmtp:97 mode-set=0,2,5,7; mode-change-period=2; \
mode-change-capability=2; mode-change-neighbor=1
a=rtpmap:98 AMR/8000/1
a=fmtp:98 mode-set=0,2,3,6; mode-change-period=2; \
mode-change-capability=2; mode-change-neighbor=1
a=rtpmap:99 AMR/8000/1
a=fmtp:99 mode-set=0,2,3,4; mode-change-period=2; \
mode-change-capability=2; mode-change-neighbor=1
a=maxptime:20
Answer:
m=audio 49120 RTP/AVP 98 99
a=rtpmap:98 AMR/8000/1
a=fmtp:98 mode-set=0,2,3,6; mode-change-period=2; \<
mode-change-capability=2; mode-change-neighbor=1
a=rtpmap:99 AMR/8000/1
a=fmtp:99 mode-set=0,2,3,4; mode-change-period=2; \
mode-change-capability=2; mode-change-neighbor=1
a=maxptime:20
The following example shows the usage of AMR between a non-GSM
endpoint and a GSM gateway. The non-GSM offerer requires no
restrictions of the mode-change-period or mode-change-neighbor, but
must signal its mode-change-capability in the offer and abide by
those restrictions in the answer.
Offer:
m=audio 49120 RTP/AVP 97
a=rtpmap:97 AMR/8000/1
a=fmtp:97 mode-change-capability=2
a=maxptime:20
Answer:
m=audio 49120 RTP/AVP 97
a=rtpmap:97 AMR/8000/1
a=fmtp:97 mode-set=0,2,4,7; mode-change-period=2; \
mode-change-capability=2; mode-change-neighbor=1
a=maxptime:20
Example of usage of AMR-WB in a possible VoIP scenario where UEP may
be used (99) and a fallback declaration (98):
m=audio 49120 RTP/AVP 99 98
a=rtpmap:98 AMR-WB/16000
a=fmtp:98 octet-align=1; mode-change-capability=2
a=rtpmap:99 AMR-WB/16000
a=fmtp:99 octet-align=1; crc=1; mode-change-capability=2
Example of usage of AMR-WB in a possible streaming scenario (two
channel stereo):
m=audio 49120 RTP/AVP 99
a=rtpmap:99 AMR-WB/16000/2
a=fmtp:99 interleaving=30
a=maxptime:100
Note that the payload format (encoding) names are commonly shown in
upper case. MIME subtypes are commonly shown in lower case. These
names are case-insensitive in both places. Similarly, parameter
names are case-insensitive both in MIME types and in the default
mapping to the SDP a=fmtp attribute.
9. IANA Considerations
Two media types (audio/AMR and audio/AMR-WB) have been updated; see
Section 8.
10. Changes from RFC 3267
The differences between RFC 3267 and this document are as follows:
- Added clarification of behavior in regards to mode change period
and mode-change neighbor that is expected from an IP client; see
Section 4.5.
- Updated the maxptime for better clarification. The sentence that
previously read: "The time SHOULD be a multiple of the frame
size." now says "The time SHOULD be an integer multiple of the
frame size." This should have no impact on interoperability.
- Updated the definition of the mode-set parameter for
clarification.
- Restricted the values for mode-change-period to 1 or 2, which are
the values used in circuit-switched AMR systems.
- Added a new media type parameter Mode-Change-Capability that
defaults to 1, which is the assumed behavior of any non-updated
implementation. This enables the offer-answer procedures to work.
- Changed mode-change-neighbor to indicate a recommended behavior
rather than a required one.
- Added an Offer-Answer Section, see Section 8.3.1. This will have
implications on the interoperability to implementations that have
guessed how to perform offer/answer negotiation of the payload
parameters.
- Clarified and aligned the unequal detection usage with the
published UDP-Lite specification in Sections 3.6.1 and 4.4.2.1.
This included replacing a normative statement about packet
handling with an informative paragraph with a reference to UDP-
Lite.
- Clarified the bit order in the CRC calculation in Section 4.4.2.1.
- Corrected the reference in Section 5.3 for the Q and FT fields.
- Changed the padding bit definition in Sections 4.4.2 and 5.3 so
that it is clear that they shall be ignored.
- Added a clarification that comfort noise frames with frame type 9,
10, and 11 SHALL NOT be used in the AMR file format.
- Clarified in Section 4.3.2 that the rules about not sending
NO_DATA frames do apply for all payload format configurations with
the exception of the interleaved mode.
- The reference list has been updated to now published RFCs: RFC
3448, RFC 3550, RFC 3551, RFC 3711, RFC 3828, and RFC 4566. A
reference to 3GPP TS 26.101 has also been added.
- Added notes in storage format section and media type registration
that AMR and AMR-WB frames can also be stored in the 3GP file
format.
- Added a media type parameter "max-red" that allows the sender to
declare a bounded usage of redundancy. This parameter allows a
receiver to optimize its function as it will know if redundancy
will be used or not. If it is used, the maximum extra delay
introduced by the sender (that is needed to be considered by the
receiver to fully utilize the redundancy) will be known. The
addition of this parameter should have no negative effects on
older implementations as they are mandated to ignore unknown
parameters per RFC 3267. In addition, older implementations are
required to operate as if the value of max-red is unknown and
possibly infinite.
- Updated the media type registration to comply with the new
registration rules.
- Moved section on decoding validation from Security Considerations
to Implementation Considerations, where it makes more sense.
- Clarified the application of encryption, integrity protection, and
authentication mechanism to the payload.
11. Acknowledgements
The authors would like to thank Petri Koskelainen, Bernhard Wimmer,
Tim Fingscheidt, Sanjay Gupta, Stephen Casner, and Colin Perkins for
their significant contributions made throughout the writing and
reviewing of RFC 3267 and this replacement. The authors would also
like to thank Richard Ejzak, Thomas Belling, and Gorry Fairhurst for
their input on this replacement of RFC 3267.
12. References
12.1. Normative References
[1] 3GPP TS 26.090, "Adaptive Multi-Rate (AMR) speech transcoding",
version 4.0.0 (2001-03), 3rd Generation Partnership Project
(3GPP).
[2] 3GPP TS 26.101, "AMR Speech Codec Frame Structure", version
4.1.0 (2001-06), 3rd Generation Partnership Project (3GPP).
[3] 3GPP TS 26.190 "AMR Wideband speech codec; Transcoding
functions", version 5.0.0 (2001-03), 3rd Generation Partnership
Project (3GPP).
[4] 3GPP TS 26.201 "AMR Wideband speech codec; Frame Structure",
version 5.0.0 (2001-03), 3rd Generation Partnership Project
(3GPP).
[5] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[6] 3GPP TS 26.093, "AMR Speech Codec; Source Controlled Rate
operation", version 4.0.0 (2000-12), 3rd Generation Partnership
Project (3GPP).
[7] 3GPP TS 26.193 "AMR Wideband Speech Codec; Source Controlled Rate operation", version 5.0.0 (2001-03), 3rd Generation Partnership Project (3GPP). [8] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. [9] 3GPP TS 26.092, "AMR Speech Codec; Comfort noise aspects", version 4.0.0 (2001-03), 3rd Generation Partnership Project (3GPP). [10] 3GPP TS 26.192 "AMR Wideband speech codec; Comfort Noise aspects", version 5.0.0 (2001-03), 3rd Generation Partnership Project (3GPP). [11] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006. [12] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, July 2003. [13] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, June 2002. [14] Freed, N. and J. Klensin, "Media Type Specifications and Registration Procedures", BCP 13, RFC 4288, December 2005. [15] Casner, S., "Media Type Registration of RTP Payload Formats", RFC 4855, February 2007.12.2. Informative References
[16] GSM 06.60, "Enhanced Full Rate (EFR) speech transcoding", version 8.0.1 (2000-11), European Telecommunications Standards Institute (ETSI). [17] ANSI/TIA/EIA-136-Rev.C, part 410 - "TDMA Cellular/PCS Radio Interface, Enhanced Full Rate Voice Codec (ACELP)". Formerly IS-641. TIA published standard, June 1 2001. [18] ARIB, RCR STD-27H, "Personal Digital Cellular Telecommunication System RCR Standard", Association of Radio Industries and Businesses (ARIB). [19] Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., and G. Fairhurst, "The Lightweight User Datagram Protocol (UDP-Lite)", RFC 3828, July 2004.
[20] 3GPP TS 25.415 "UTRAN Iu Interface User Plane Protocols", version 4.2.0 (2001-09), 3rd Generation Partnership Project (3GPP). [21] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP Friendly Rate Control (TFRC): Protocol Specification", RFC 3448, January 2003. [22] Li, A., et al., "An RTP Payload Format for Generic FEC with Uneven Level Protection", Work in Progress. [23] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for Generic Forward Error Correction", RFC 2733, December 1999. [24] 3GPP TS 26.102, "AMR speech codec interface to Iu and Uu", version 4.0.0 (2001-03), 3rd Generation Partnership Project (3GPP). [25] 3GPP TS 26.202, "AMR Wideband speech codec; Interface to Iu and Uu", version 5.0.0 (2001-03), 3rd Generation Partnership Project (3GPP). [26] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, March 2004. [27] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, September 1997. [28] 3GPP TS 26.103, "Speech codec list for GSM and UMTS", version 5.5.0 (2004-09), 3rd Generation Partnership Project (3GPP). [29] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming Protocol (RTSP)", RFC 2326, April 1998. [30] Handley, M., Perkins, C., and E. Whelan, "Session Announcement Protocol", RFC 2974, October 2000. [31] 3GPP TS 26.244, "3GPP file format (3GP)", version 6.1.0 (2004- 09), 3rd Generation Partnership Project (3GPP). [32] Castagno, R. and D. Singer, "MIME Type Registrations for 3rd Generation Partnership Project (3GPP) Multimedia files", RFC 3839, July 2004. [33] Kent, S. and K. Seo, "Security Architecture for the Internet Protocol", RFC 4301, December 2005.
[34] Dierks, T. and E. Rescorla, "The Transport Layer Security (TLS) Protocol Version 1.1", RFC 4346, April 2006. ETSI documents are available from <http://www.etsi.org/>. 3GPP documents are available from <http://www.3gpp.org/>. TIA documents are available from <http://www.tiaonline.org/>.Authors' Addresses
Johan Sjoberg Ericsson AB SE-164 80 Stockholm, SWEDEN Phone: +46 8 7190000 EMail: Johan.Sjoberg@ericsson.com Magnus Westerlund Ericsson Research Ericsson AB SE-164 80 Stockholm, SWEDEN Phone: +46 8 7190000 EMail: Magnus.Westerlund@ericsson.com Ari Lakaniemi Nokia Research Center P.O.Box 407 FIN-00045 Nokia Group, FINLAND Phone: +358-71-8008000 EMail: ari.lakaniemi@nokia.com Qiaobing Xie Motorola, Inc. 1501 W. Shure Drive, 2-B8 Arlington Heights, IL 60004, USA Phone: +1-847-632-3028 EMail: Qiaobing.Xie@motorola.com
Full Copyright Statement Copyright (C) The IETF Trust (2007). This document is subject to the rights, licenses and restrictions contained in BCP 78, and except as set forth therein, the authors retain all their rights. This document and the information contained herein are provided on an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST AND THE INTERNET ENGINEERING TASK FORCE DISCLAIM 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. Intellectual Property The IETF takes no position regarding the validity or scope of any Intellectual Property Rights 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; nor does it represent that it has made any independent effort to identify any such rights. Information on the procedures with respect to rights in RFC documents can be found in BCP 78 and BCP 79. Copies of IPR disclosures made to the IETF Secretariat 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 implementers or users of this specification can be obtained from the IETF on-line IPR repository at http://www.ietf.org/ipr. The IETF invites any interested party to bring to its attention any copyrights, patents or patent applications, or other proprietary rights that may cover technology that may be required to implement this standard. Please address the information to the IETF at <%ietf-ipr@ietf.org. Acknowledgement Funding for the RFC Editor function is currently provided by the Internet Society.