Internet DRAFT - draft-ietf-avt-rtp-amr-bis
draft-ietf-avt-rtp-amr-bis
Network Working Group Johan Sjoberg
INTERNET-DRAFT Magnus Westerlund
Expires: March 2007 Ericsson
Obsoletes(if approved): RFC 3267 Ari Lakaniemi
Nokia
Q. Xie
Motorola
September 18, 2006
RTP Payload Format and File Storage Format for the Adaptive Multi-
Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs
<draft-ietf-avt-rtp-amr-bis-06.txt>
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Abstract
This document specifies a real-time transport protocol (RTP) payload
format to be used for Adaptive Multi-Rate (AMR) and Adaptive
Multi-Rate Wideband (AMR-WB) encoded speech signals. The payload
format is designed to be able to interoperate with existing AMR and
AMR-WB transport formats on non-IP networks. In addition, a file
format is specified for transport of AMR and AMR-WB speech data in
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storage mode applications such as email. Two separate media type
registrations are included, one for AMR and one for AMR-WB,
specifying use of both the RTP payload format and the storage
format. This document obsoletes RFC 3267.
Table of Contents
1. Introduction.....................................................3
2. Conventions and Acronyms.........................................4
3. Background on AMR/AMR-WB and Design Principles...................4
3.1. The Adaptive Multi-Rate (AMR) Speech Codec..................5
3.2. The Adaptive Multi-Rate Wideband (AMR-WB) Speech Codec......5
3.3. Multi-rate Encoding and Mode Adaptation.....................5
3.4. Voice Activity Detection and Discontinuous Transmission.....6
3.5. Support for Multi-Channel Session...........................6
3.6. Unequal Bit-error Detection and Protection..................7
3.6.1. Applying UEP and UED in an IP Network..................8
3.7. Robustness against Packet Loss..............................9
3.7.1. Use of Forward Error Correction (FEC)..................9
3.7.2. Use of Frame Interleaving.............................11
3.8. Bandwidth Efficient or Octet-aligned Mode..................11
3.9. AMR or AMR-WB Speech over IP scenarios.....................12
4. AMR and AMR-WB RTP Payload Formats..............................14
4.1. RTP Header Usage...........................................14
4.2. Payload Structure..........................................16
4.3. Bandwidth-Efficient Mode...................................16
4.3.1. The Payload Header....................................16
4.3.2. The Payload Table of Contents.........................17
4.3.3. Speech Data...........................................19
4.3.4. Algorithm for Forming the Payload.....................21
4.3.5. Payload Examples......................................22
4.3.5.1. Single Channel Payload Carrying a Single Frame...22
4.3.5.2. Single Channel Payload Carrying Multiple Frames..23
4.3.5.3. Multi-Channel Payload Carrying Multiple Frames...24
4.4. Octet-aligned Mode.........................................25
4.4.1. The Payload Header....................................25
4.4.2. The Payload Table of Contents and Frame CRCs..........26
4.4.2.1. Use of Frame CRC for UED over IP.................28
4.4.3. Speech Data...........................................29
4.4.4. Methods for Forming the Payload.......................30
4.4.5. Payload Examples......................................32
4.4.5.1. Basic Single Channel Payload Carrying Multiple
Frames...........................................32
4.4.5.2. Two Channel Payload with CRC, Interleaving, and
Robust-sorting...................................32
4.5. Implementation Considerations..............................33
4.5.1. Decoding Validation...................................34
5. AMR and AMR-WB Storage Format...................................35
5.1. Single channel Header......................................36
5.2. Multi-channel Header.......................................36
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5.3. Speech Frames..............................................37
6. Congestion Control..............................................38
7. Security Considerations.........................................39
7.1. Confidentiality............................................39
7.2. Authentication and Integrity...............................39
8. Payload Format Parameters.......................................40
8.1. AMR Media Type Registration................................40
8.2. AMR-WB Media Type Registration.............................44
8.3. Mapping Media Type Parameters into SDP.....................47
8.3.1. Offer-Answer Model Considerations.....................47
8.3.2. Usage of declarative SDP..............................50
8.3.3. Examples..............................................51
9. IANA Considerations.............................................52
10. Changes........................................................53
11. Acknowledgements...............................................54
12. References.....................................................55
12.1. Normative References......................................55
12.2. Informative References....................................56
13. Authors' Addresses.............................................57
14. IPR Notice.....................................................58
15. Copyright Notice...............................................58
1. Introduction
This document obsoletes RFC 3267 and extends that specification with
offer/answer rules. See Section 10 for the changes made to this
format in relation to RFC 3267.
This document specifies the payload format for packetization of AMR
and AMR-WB encoded speech signals into the Real-time Transport
Protocol (RTP)[8]. The payload format supports transmission of
multiple channels, multiple frames per payload, the use of fast
codec mode adaptation, robustness against packet loss and bit
errors, and interoperation with existing AMR and AMR-WB transport
formats on non-IP networks, as described in Section 3.
The payload format itself is specified in Section 4. A related file
format is specified in Section 5 for transport of AMR and AMR-WB
speech data in storage mode applications such as email. In Section
8, two separate media type registrations are provided, one for AMR
and one for AMR-WB.
Even though this RTP payload format definition supports the
transport of both AMR and AMR-WB speech, it is important to remember
that AMR and AMR-WB are two different codecs and they are always
handled as different payload types in RTP.
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2. Conventions and Acronyms
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC2119 [5].
The following acronyms are used in this document:
3GPP - the Third Generation Partnership Project
AMR - Adaptive Multi-Rate (Codec)
AMR-WB - Adaptive Multi-Rate Wideband (Codec)
CMR - Codec Mode Request
CN - Comfort Noise
DTX - Discontinuous Transmission
ETSI - European Telecommunications Standards Institute
FEC - Forward Error Correction
SCR - Source Controlled Rate Operation
SID - Silence Indicator (the frames containing only CN
parameters)
VAD - Voice Activity Detection
UED - Unequal Error Detection
UEP - Unequal Error Protection
The term "frame-block" is used in this document to describe the
time-synchronized set of speech frames in a multi-channel AMR or
AMR-WB session. In particular, in an N-channel session, a
frame- block will contain N speech frames, one from each of the
channels, and all N speech frames represents exactly the same time
period.
3. Background on AMR/AMR-WB and Design Principles
AMR and AMR-WB were originally designed for circuit-switched mobile
radio systems. Due to their flexibility and robustness, they are
also suitable for other real-time speech communication services over
packet-switched networks such as the Internet.
Because of the flexibility of these codecs, the behavior in a
particular application is controlled by several parameters that
select options or specify the acceptable values for a variable.
These options and variables are described in general terms at
appropriate points in the text of this specification as parameters
to be established through out-of-band means. In Section 8, all of
the parameters are specified in the form of media subtype
registrations for the AMR and AMR-WB encodings. The method used to
signal these parameters at session setup or to arrange prior
agreement of the participants is beyond the scope of this document;
however, Section 8.3 provides a mapping of the parameters into the
Session Description Protocol (SDP) [11] for those applications that
use SDP.
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3.1. The Adaptive Multi-Rate (AMR) Speech Codec
The AMR codec was originally developed and standardized by the
European Telecommunications Standards Institute (ETSI) for GSM
cellular systems. It is now chosen by the Third Generation
Partnership Project (3GPP) as the mandatory codec for third
generation (3G) cellular systems [1].
The AMR codec is a multi-mode codec that supports eight narrow band
speech encoding modes with bit rates between 4.75 and 12.2 kbps.
The sampling frequency used in AMR is 8000 Hz and the speech
encoding is performed on 20 ms speech frames. Therefore, each
encoded AMR speech frame represents 160 samples of the original
speech.
Among the eight AMR encoding modes, three are already separately
adopted as standards of their own. Particularly, the 6.7 kbps mode
is adopted as PDC-EFR [18], the 7.4 kbps mode as IS-641 codec in
TDMA [17], and the 12.2 kbps mode as GSM-EFR [16].
3.2. The Adaptive Multi-Rate Wideband (AMR-WB) Speech Codec
The Adaptive Multi-Rate Wideband (AMR-WB) speech codec [3] was
originally developed by 3GPP to be used in GSM and 3G cellular
systems.
Similar to AMR, the AMR-WB codec is also a multi-mode speech codec.
AMR-WB supports nine wide band speech coding modes with respective
bit rates ranging from 6.6 to 23.85 kbps. The sampling frequency
used in AMR-WB is 16000 Hz and the speech processing is performed on
20 ms frames. This means that each AMR-WB encoded frame represents
320 speech samples.
3.3. Multi-rate Encoding and Mode Adaptation
The multi-rate encoding (i.e., multi-mode) capability of AMR and
AMR-WB is designed for preserving high speech quality under a wide
range of transmission conditions.
With AMR or AMR-WB, mobile radio systems are able to use available
bandwidth as effectively as possible. E.g., in GSM it is possible
to dynamically adjust the speech encoding rate during a session so
as to continuously adapt to the varying transmission conditions by
dividing the fixed overall bandwidth between speech data and error
protective coding to enable best possible trade-off between speech
compression rate and error tolerance. To perform mode adaptation,
the decoder (speech receiver) needs to signal the encoder (speech
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sender) the new mode it prefers. This mode change signal is called
Codec Mode Request or CMR.
Since in most sessions speech is sent in both directions between the
two ends, the mode requests from the decoder at one end to the
encoder at the other end are piggy-backed over the speech frames in
the reverse direction. In other words, there is no out-of-band
signaling needed for sending CMRs.
Every AMR or AMR-WB codec implementation is required to support all
the respective speech coding modes defined by the codec and must be
able to handle mode switching to any of the modes at any time.
However, some transport systems may impose limitations in the number
of modes supported and how often the mode can change due to
bandwidth limitations or other constraints. For this reason, the
decoder is allowed to indicate its acceptance of a particular mode
or a subset of the defined modes for the session using out-of-band
means.
For example, the GSM radio link can only use a subset of at most
four different modes in a given session. This subset can be any
combination of the eight AMR modes for an AMR session or any
combination of the nine AMR-WB modes for an AMR-WB session.
Moreover, for better interoperability with GSM through a gateway,
the decoder is allowed to use out-of-band means to set the minimum
number of frames between two mode changes and to limit the mode
change among neighboring modes only.
Section 8 specifies a set of media type parameters that may be used
to signal these mode adaptation controls at session setup.
3.4. Voice Activity Detection and Discontinuous Transmission
Both codecs support voice activity detection (VAD) and generation of
comfort noise (CN) parameters during silence periods. Hence, the
codecs have the option to reduce the number of transmitted bits and
packets during silence periods to a minimum. The operation of
sending CN parameters at regular intervals during silence periods is
usually called discontinuous transmission (DTX) or source controlled
rate (SCR) operation. The AMR or AMR-WB frames containing CN
parameters are called Silence Indicator (SID) frames. See more
details about VAD and DTX functionality in [9] and [10].
3.5. Support for Multi-Channel Session
Both the RTP payload format and the storage format defined in this
document support multi-channel audio content (e.g., a stereophonic
speech session).
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Although AMR and AMR-WB codecs themselves do not support encoding of
multi-channel audio content into a single bit stream, they can be
used to separately encode and decode each of the individual
channels.
To transport (or store) the separately encoded multi-channel
content, the speech frames for all channels that are framed and
encoded for the same 20 ms periods are logically collected in a
frame-block.
At the session setup, out-of-band signaling must be used to indicate
the number of channels in the session and the order of the speech
frames from different channels in each frame-block. When using SDP
for signaling, the number of channels is specified in the rtpmap
attribute and the order of channels carried in each frame-block is
implied by the number of channels as specified in Section 4.1 in
[12].
3.6. Unequal Bit-error Detection and Protection
The speech bits encoded in each AMR or AMR-WB frame have different
perceptual sensitivity to bit errors. This property has been
exploited in cellular systems to achieve better voice quality by
using unequal error protection and detection (UEP and UED)
mechanisms.
The UEP/UED mechanisms focus the protection and detection of
corrupted bits to the perceptually most sensitive bits in an AMR or
AMR-WB frame. In particular, speech bits in an AMR or AMR-WB frame
are divided into class A, B, and C, where bits in class A are most
sensitive and bits in class C least sensitive (see Table 1 below for
AMR and [4] for AMR-WB). An AMR or AMR-WB frame is only declared
damaged if there are bit errors found in the most sensitive bits,
i.e., the class A bits. On the other hand, it is acceptable to have
some bit errors in the other bits, i.e., class B and C bits.
Class A total speech
Index Mode bits bits
----------------------------------------
0 AMR 4.75 42 95
1 AMR 5.15 49 103
2 AMR 5.9 55 118
3 AMR 6.7 58 134
4 AMR 7.4 61 148
5 AMR 7.95 75 159
6 AMR 10.2 65 204
7 AMR 12.2 81 244
8 AMR SID 39 39
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Table 1. The number of class A bits for the AMR codec.
Moreover, a damaged frame is still useful for error concealment at
the decoder since some of the less sensitive bits can still be used.
This approach can improve the speech quality compared to discarding
the damaged frame.
3.6.1. Applying UEP and UED in an IP Network
To take full advantage of the bit-error robustness of the AMR and
AMR-WB codec, the RTP payload format is designed to facilitate
UEP/UED in an IP network. It should be noted however that the
utilization of UEP and UED discussed below is OPTIONAL.
UEP/UED in an IP network can be achieved by detecting bit errors in
class A bits and tolerating bit errors in class B/C bits of the AMR
or AMR-WB frame(s) in each RTP payload.
Link layer protocols exist that do not discard packets containing
bit errors, e.g., SLIP and some wireless links. With the Internet
traffic pattern shifting towards a more multimedia-centric one, more
link layers of such nature may emerge in the future. With transport
layer support for partial checksums, for example those supported by
UDP-Lite [19], bit error tolerant AMR and AMR-WB traffic could
achieve better performance over these types of links. The
relationship between UDP-Lite's partial checksum at the Transport
Layer and the checksum coverage provided by the link-layer frame is
described in UDP-Lite specification [19].
There are at least two basic approaches for carrying AMR and AMR-WB
traffic over bit error tolerant IP networks:
a) Utilizing a partial checksum to cover the IP, transport protocol
(e.g. UDP-Lite), RTP and payload headers, and the most important
speech bits of the payload. The IP, UDP and RTP headers need to
be protected, and it is recommended that at least all class A
bits are covered by the checksum.
b) Utilizing a partial checksum to only cover the IP, transport
protocol, RTP and payload headers, but an AMR or AMR-WB frame CRC
to cover the class A bits of each speech frame in the RTP
payload.
In either approach, at least part of the class B/C bits are left
without error-check and thus bit error tolerance is achieved.
Note, it is still important that the network designer pay
attention to the class B and C residual bit error rate. Though
less sensitive to errors than class A bits, class B and C bits
are not insignificant and undetected errors in these bits cause
degradation in speech quality. An example of residual error
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rates considered acceptable for AMR in UMTS can be found in [24]
and for AMR-WB in [25].
The application interface to the UEP/UED transport protocol (e.g.,
UDP-Lite) may not provide any control over the link error rate,
especially in a gateway scenario. Therefore, it is incumbent upon
the designer of a node with a link interface of this type to choose
a residual bit error rate that is low enough to support applications
such as AMR encoding when transmitting packets of a UEP/UED
transport protocol.
Approach 1 is bit efficient, flexible and simple, but comes with two
disadvantages, namely, a) bit errors in protected speech bits will
cause the payload to be discarded, and b) when transporting multiple
AMR or AMR-WB frames in a RTP payload there is the possibility that
a single bit error in protected bits will cause all the frames to be
discarded.
These disadvantages can be avoided, if needed, with some overhead in
the form of a frame-wise CRC (Approach 2). In problem a), the CRC
makes it possible to detect bit errors in class A bits and use the
frame for error concealment, which gives a small improvement in
speech quality. For b), when transporting multiple frames in a
payload, the CRCs remove the possibility that a single bit error in
a class A bit will cause all the frames to be discarded. Avoiding
that gives an improvement in speech quality when transporting
multiple AMR or AMR-WB frames over links subject to bit errors.
The choice between the above two approaches must be made based on
the available bandwidth, and desired tolerance to bit errors.
Neither solution is appropriate to all cases. Section 8 defines
parameters that may be used at session setup to select between these
approaches.
3.7. Robustness against Packet Loss
The payload format supports several means, including forward error
correction (FEC) and frame interleaving, to increase robustness
against packet loss.
3.7.1. Use of Forward Error Correction (FEC)
The simple scheme of repetition of previously sent data is one way
of achieving FEC. Another possible scheme which is more bandwidth
efficient is to use payload external FEC, e.g., RFC2733 [23], which
generates extra packets containing repair data. The whole payload
can also be sorted in sensitivity order to support external FEC
schemes using UEP. There is also a work in progress on a generic
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version of such a scheme [22] that can be applied to AMR or AMR-WB
payload transport.
With AMR or AMR-WB, it is possible to use the multi-rate capability
of the codec to send redundant copies of the same mode or of another
mode, e.g., one with lower-bandwidth. We describe such a scheme
next.
This involves the simple retransmission of previously transmitted
frame-blocks together with the current frame-block(s). This is done
by using a sliding window to group the speech frame-blocks to send
in each payload. Figure 1 below shows us an example.
--+--------+--------+--------+--------+--------+--------+--------+--
| f(n-2) | f(n-1) | f(n) | f(n+1) | f(n+2) | f(n+3) | f(n+4) |
--+--------+--------+--------+--------+--------+--------+--------+--
<---- p(n-1) ---->
<----- p(n) ----->
<---- p(n+1) ---->
<---- p(n+2) ---->
<---- p(n+3) ---->
<---- p(n+4) ---->
Figure 1: An example of redundant transmission.
In this example each frame-block is retransmitted one time in the
following RTP payload packet. Here, f(n-2)..f(n+4) denotes a
sequence of speech frame-blocks and p(n-1)..p(n+4) a sequence of
payload packets.
The use of this approach does not require signaling at the session
setup. However a parameter for providing a maximum delay in
transmitting any redundant frame is defined. In other words, the
speech sender can choose to use this scheme without consulting the
receiver. This is because a packet containing redundant frames will
not look different from a packet with only new frames. The receiver
may receive multiple copies or versions (encoded with different
modes) of a frame for a certain timestamp if no packet is lost. If
multiple versions of the same speech frame are received, it is
recommended that the mode with the highest rate be used by the
speech decoder.
This redundancy scheme provides the same functionality as the one
described in RFC 2198 "RTP Payload for Redundant Audio Data" [27].
In most cases the mechanism in this payload format is more efficient
and simpler than requiring both endpoints to support RFC 2198 in
addition. There are two situations in which use of RFC 2198 is
indicated: if the spread in time required between the primary and
redundant encodings is larger than 5 frame times, the bandwidth
overhead of RFC 2198 will be lower; or, if a non-AMR codec is
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desired for the redundant encoding, the AMR payload format won't be
able to carry it.
The sender is responsible for selecting an appropriate amount of
redundancy based on feedback about the channel, e.g., in RTCP
receiver reports. A sender should not base selection of FEC on the
CMR, as this parameter most probably was set based on none-IP
information, e.g., radio link performance measures. The sender is
also responsible for avoiding congestion, which may be exacerbated
by redundancy (see Section 6 for more details).
3.7.2. Use of Frame Interleaving
To decrease protocol overhead, the payload design allows several
speech frame-blocks be encapsulated into a single RTP packet. One
of the drawbacks of such an approach is that in case of packet loss
this means loss of several consecutive speech frame-blocks, which
usually causes clearly audible distortion in the reconstructed
speech. Interleaving of frame-blocks can improve the speech quality
in such cases by distributing the consecutive losses into a series
of single frame-block losses. However, interleaving and bundling
several frame-blocks per payload will also increase end-to-end delay
and is therefore not appropriate for all types of applications.
Streaming applications will most likely be able to exploit
interleaving to improve speech quality in lossy transmission
conditions.
This payload design supports the use of frame interleaving as an
option. For the encoder (speech sender) to use frame interleaving
in its outbound RTP packets for a given session, the decoder (speech
receiver) needs to indicate its support via out-of-band means (see
Section 8).
3.8. Bandwidth Efficient or Octet-aligned Mode
For a given session, the payload format can be either bandwidth
efficient or octet aligned, depending on the mode of operation that
is established for the session via out-of-band means.
In the octet-aligned format, all the fields in a payload, including
payload header, table of contents entries, and speech frames
themselves, are individually aligned to octet boundaries to make
implementations efficient. In the bandwidth efficient format only
the full payload is octet aligned, so fewer padding bits are added.
Note, octet alignment of a field or payload means that the last
octet is padded with zeroes in the least significant bits to fill
the octet. Also note that this padding is separate from padding
indicated by the P bit in the RTP header.
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Between the two operation modes, only the octet-aligned mode has the
capability to use the robust sorting, interleaving, and frame CRC to
make the speech transport more robust to packet loss and bit errors.
3.9. AMR or AMR-WB Speech over IP scenarios
The primary scenario for this payload format is IP end-to-end
between two terminals, as shown in Figure 2. This payload format is
expected to be useful for both conversational and streaming
services.
+----------+ +----------+
| | IP/UDP/RTP/AMR or | |
| TERMINAL |<----------------------->| TERMINAL |
| | IP/UDP/RTP/AMR-WB | |
+----------+ +----------+
Figure 2: IP terminal to IP terminal scenario
A conversational service puts requirements on the payload format.
Low delay is one very important factor, i.e., few speech
frame-blocks per payload packet. Low overhead is also required when
the payload format traverses low bandwidth links, especially as the
frequency of packets will be high. For low bandwidth links it also
an advantage to support UED which allows a link provider to reduce
delay and packet loss or to reduce the utilization of link
resources.
A Streaming service has less strict real-time requirements and
therefore can use a larger number of frame-blocks per packet than a
conversational service. This reduces the overhead from IP, UDP, and
RTP headers. However, including several frame-blocks per packet
makes the transmission more vulnerable to packet loss, so
interleaving may be used to reduce the effect packet loss will have
on speech quality. A streaming server handling a large number of
clients also needs a payload format that requires as few resources
as possible when doing packetization. The octet-aligned and
interleaving modes require the least amount of resources, while CRC,
robust sorting, and bandwidth efficient modes have higher demands.
Another scenario occurs when AMR or AMR-WB encoded speech will be
transmitted from a non-IP system (e.g., a GSM or 3GPP UMTS network)
to an IP/UDP/RTP VoIP terminal, and/or vice versa, as depicted in
Figure 3.
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AMR or AMR-WB
over
I.366.{2,3} or +------+ +----------+
3G Iu or | | IP/UDP/RTP/AMR or | |
<------------->| GW |<---------------------->| TERMINAL |
GSM Abis | | IP/UDP/RTP/AMR-WB | |
etc. +------+ +----------+
|
GSM/ | IP network
3GPP UMTS network |
Figure 3: GW to VoIP terminal scenario
In such a case, it is likely that the AMR or AMR-WB frame is
packetized in a different way in the non-IP network and will need to
be re-packetized into RTP at the gateway. Also, speech frames from
the non-IP network may come with some UEP/UED information (e.g., a
frame quality indicator) that will need to be preserved and
forwarded on to the decoder along with the speech bits. This is
specified in Section 4.3.2.
AMR's capability to do fast mode switching is exploited in some
non-IP networks to optimize speech quality. To preserve this
functionality in scenarios including a gateway to an IP network, a
codec mode request (CMR) field is needed. The gateway will be
responsible for forwarding the CMR between the non-IP and IP parts
in both directions. The IP terminal should follow the CMR forwarded
by the gateway to optimize speech quality going to the non-IP
decoder. The mode control algorithm in the gateway must accommodate
the delay imposed by the IP network on the response to CMR by the IP
terminal.
The IP terminal should not set the CMR (see Section 4.3.1), but the
gateway can set the CMR value on frames going toward the encoder in
the non-IP part to optimize speech quality from that encoder to the
gateway. The gateway can alternatively set a lower CMR value, if
desired, as one means to control congestion on the IP network.
A third likely scenario is that IP/UDP/RTP is used as transport
between two non-IP systems, i.e., IP is originated and terminated in
gateways on both sides of the IP transport, as illustrated in Figure
4 below.
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AMR or AMR-WB AMR or AMR-WB
over over
I.366.{2,3} or +------+ +------+ I.366.{2,3} or
3G Iu or | | IP/UDP/RTP/AMR or | | 3G Iu or
<------------->| GW |<------------------->| GW |<------------->
GSM Abis | | IP/UDP/RTP/AMR-WB | | GSM Abis
etc. +------+ +------+ etc.
| |
GSM/ | IP network | GSM/
3GPP UMTS network | | 3GPP UMTS network
Figure 4: GW to GW scenario
This scenario requires the same mechanisms for preserving UED/UEP
and CMR information as in the single gateway scenario. In addition,
the CMR value may be set in packets received by the gateways on the
IP network side. The gateway should forward to the non-IP side a
CMR value that is the minimum of three values:
- the CMR value it receives on the IP side;
- the CMR value it calculates based on its reception quality on
the non-IP side; and
- a CMR value it may choose for congestion control of
transmission on the IP side.
The details of the control algorithm are left to the implementation.
4. AMR and AMR-WB RTP Payload Formats
The AMR and AMR-WB payload formats have identical structure, so they
are specified together. The only differences are in the types of
codec frames contained in the payload. The payload format consists
of the RTP header, payload header and payload data.
4.1. RTP Header Usage
The format of the RTP header is specified in [8]. This payload
format uses the fields of the header in a manner consistent with
that specification.
The RTP timestamp corresponds to the sampling instant of the first
sample encoded for the first frame-block in the packet. The
timestamp clock frequency is the same as the sampling frequency, so
the timestamp unit is in samples.
The duration of one speech frame-block is 20 ms for both AMR and
AMR-WB. For AMR, the sampling frequency is 8 kHz, corresponding to
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160 encoded speech samples per frame from each channel. For AMR-WB,
the sampling frequency is 16 kHz, corresponding to 320 samples per
frame from each channel. Thus, the timestamp is increased by 160
for AMR and 320 for AMR-WB for each consecutive frame-block.
A packet may contain multiple frame-blocks of encoded speech or
comfort noise parameters. If interleaving is employed, the
frame-blocks encapsulated into a payload are picked according to the
interleaving rules as defined in Section 4.4.1. Otherwise, each
packet covers a period of one or more contiguous 20 ms frame-block
intervals. In case the data from all the channels for a particular
frame-block in the period is missing, for example at a gateway from
some other transport format, it is possible to indicate that no data
is present for that frame-block rather than breaking a
multi-frame-block packet into two, as explained in Section 4.3.2.
To allow for error resiliency through redundant transmission, the
periods covered by multiple packets MAY overlap in time. A receiver
MUST be prepared to receive any speech frame multiple times, either
in exact duplicates, or in different AMR rate modes, or with data
present in one packet and not present in another. If multiple
versions of the same speech frame are received, it is RECOMMENDED
that the mode with the highest rate be used by the speech decoder.
A given frame MUST NOT be encoded as speech in one packet and
comfort noise parameters in another.
The payload is always made an integral number of octets long by
padding with zero bits if necessary. If additional padding is
required to bring the payload length to a larger multiple of octets
or for some other purpose, then the P bit in the RTP in the header
may be set and padding appended as specified in [8].
The RTP header marker bit (M) SHALL be set to 1 if the first
frame-block carried in the packet contains a speech frame which is
the first in a talkspurt. For all other packets the marker bit
SHALL be set to zero (M=0).
The assignment of an RTP payload type for this new packet format is
outside the scope of this document, and will not be specified here.
It is expected that the RTP profile under which this payload format
is being used will assign a payload type for this encoding or
specify that the payload type is to be bound dynamically.
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4.2. Payload Structure
The complete payload consists of a payload header, a payload table
of contents, and speech data representing one or more speech
frame-blocks. The following diagram shows the general payload
format layout:
+----------------+-------------------+----------------
| payload header | table of contents | speech data ...
+----------------+-------------------+----------------
Payloads containing more than one speech frame-block are called
compound payloads.
The following sections describe the variations taken by the payload
format depending on whether the AMR session is set up to use the
bandwidth-efficient mode or octet-aligned mode and any of the
OPTIONAL functions for robust sorting, interleaving, and frame CRCs.
Implementations SHOULD support both bandwidth-efficient and
octet-aligned operation to increase interoperability.
4.3. Bandwidth-Efficient Mode
4.3.1. The Payload Header
In bandwidth-efficient mode, the payload header simply consists of a
4 bit codec mode request:
0 1 2 3
+-+-+-+-+
| CMR |
+-+-+-+-+
CMR (4 bits): Indicates a codec mode request sent to the speech
encoder at the site of the receiver of this payload. The value
of the CMR field is set to the frame type index of the
corresponding speech mode being requested. The frame type index
may be 0-7 for AMR, as defined in Table 1a in [2], or 0-8 for
AMR-WB, as defined in Table 1a in [4]. CMR value 15 indicates
that no mode request is present, and other values are for future
use.
The codec mode request received in the CMR field is valid until the
next codec mode request is received, i.e., a newly received CMR
value corresponding to a speech mode or NO_DATA overrides the
previously received CMR value corresponding to a speech mode or
NO_DATA. Therefore, if a terminal continuously wishes to receive
frames in the same mode X, it needs to set CMR=X for all its
outbound payloads, and if a terminal has no preference in which mode
to receive, it SHOULD set CMR=15 in all its outbound payloads.
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If receiving a payload with a CMR value that is not a speech mode or
NO_DATA, the CMR MUST be ignored by the receiver.
In a multi-channel session, codec mode request SHOULD be interpreted
by the receiver of the payload as the desired encoding mode for all
the channels in the session.
An IP end-point SHOULD NOT set the codec mode request based on
packet losses or other congestion indications, for several reasons:
- The other end of the IP path may be a gateway to a non-IP
network (such as a radio link) that needs to set the CMR field
to optimize performance on that network.
- Congestion on the IP network is managed by the IP sender, in
this case at the other end of the IP path. Feedback about
congestion SHOULD be provided to that IP sender through RTCP
or other means, and then the sender can choose to avoid
congestion using the most appropriate mechanism. That may
include adjusting the codec mode, but also includes adjusting
the level of redundancy or number of frames per packet.
The encoder SHOULD follow a received codec mode request, but MAY
change to a lower-numbered mode if it so chooses, for example to
control congestion.
The CMR field MUST be set to 15 for packets sent to a multicast
group. The encoder in the speech sender SHOULD ignore codec mode
requests when sending speech to a multicast session but MAY use RTCP
feedback information as a hint that a codec mode change is needed.
The codec mode selection MAY be restricted by a session parameter to
a subset of the available modes. If so, the requested mode MUST be
among the signalled subset (see Section 8).
4.3.2. The Payload Table of Contents
The table of contents (ToC) consists of a list of ToC entries, each
representing a speech frame.
In bandwidth-efficient mode, a ToC entry takes the following format:
0 1 2 3 4 5
+-+-+-+-+-+-+
|F| FT |Q|
+-+-+-+-+-+-+
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F (1 bit): If set to 1, indicates that this frame is followed by
another speech frame in this payload; if set to 0, indicates that
this frame is the last frame in this payload.
FT (4 bits): Frame type index, indicating either the AMR or AMR-WB
speech coding mode or comfort noise (SID) mode of the
corresponding frame carried in this payload.
The value of FT is defined in Table 1a in [2] for AMR and in Table
1a in [4] for AMR-WB. FT=14 (SPEECH_LOST, only available for
AMR-WB) and FT=15 (NO_DATA) are used to indicate frames that are
either lost or not being transmitted in this payload, respectively.
NO_DATA (FT=15) frame could mean either that there is no data
produced by the speech encoder for that frame or that no data for
that frame is transmitted in the current payload (i.e., valid data
for that frame could be sent in either an earlier or later packet).
If receiving a ToC entry with a FT value in the range 9-14 for AMR
or 10-13 for AMR-WB the whole packet SHOULD be discarded. This is
to avoid the loss of data synchronization in the depacketization
process, which can result in a huge degradation in speech quality.
Note that packets containing only NO_DATA frames SHOULD NOT be
transmitted, independently of payload format configuration with the
exception of interleaving. Also, frame-blocks containing only
NO_DATA frames at the end of a packet SHOULD NOT be transmitted in
any payload format configuration, except in the case of
interleaving. The AMR SCR/DTX is described in [6] and AMR-WB
SCR/DTX in [7].
The extra comfort noise frame types specified in table 1a in [2]
(i.e., GSM-EFR CN, IS-641 CN, and PDC-EFR CN) MUST NOT be used in
this payload format because the standardized AMR codec is only
required to implement the general AMR SID frame type and not those
that are native to the incorporated encodings.
Q (1 bit): Frame quality indicator. If set to 0, indicates the
corresponding frame is severely damaged and the receiver should
set the RX_TYPE (see [6]) to either SPEECH_BAD or SID_BAD
depending on the frame type (FT).
The frame quality indicator is included for interoperability with
the ATM payload format described in ITU-T I.366.2, the UMTS Iu
interface [20], as well as other transport formats. The frame
quality indicator enables damaged frames to be forwarded to the
speech decoder for error concealment. This can improve the speech
quality comparing to dropping the damaged frames. See Section
4.4.2.1 for more details.
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For multi-channel sessions, the ToC entries of all frames from a
frame-block are placed in the ToC in consecutive order as defined in
Section 4.1 in [12]. When multiple frame-blocks are present in a
packet in bandwidth-efficient mode, they will be placed in the
packet in order of their creation time.
Therefore, with N channels and K speech frame-blocks in a packet,
there MUST be N*K entries in the ToC, and the first N entries will
be from the first frame-block, the second N entries will be from the
second frame-block, and so on.
The following figure shows an example of a ToC of three entries in a
single channel session using bandwidth efficient mode.
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| FT |Q|1| FT |Q|0| FT |Q|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Below is an example of how the ToC entries will appear in the ToC of
a packet carrying 3 consecutive frame-blocks in a session with two
channels (L and R).
+----+----+----+----+----+----+
| 1L | 1R | 2L | 2R | 3L | 3R |
+----+----+----+----+----+----+
|<------->|<------->|<------->|
Frame- Frame- Frame-
Block 1 Block 2 Block 3
4.3.3. Speech Data
Speech data of a payload contains zero or more speech frames or
comfort noise frames, as described in the ToC of the payload.
Note, for ToC entries with FT=14 or 15, there will be no
corresponding speech frame present in the speech data.
Each speech frame represents 20 ms of speech encoded with the mode
indicated in the FT field of the corresponding ToC entry. The
length of the speech frame is implicitly defined by the mode
indicated in the FT field. The order and numbering notation of the
bits are as specified for Interface Format 1 (IF1) in [2] for AMR
and [4] for AMR-WB. As specified there, the bits of speech frames
have been rearranged in order of decreasing sensitivity, while the
bits of comfort noise frames are in the order produced by the
encoder. The resulting bit sequence for a frame of length K bits is
denoted d(0), d(1), ..., d(K-1).
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4.3.4. Algorithm for Forming the Payload
The complete RTP payload in bandwidth-efficient mode is formed by
packing bits from the payload header, table of contents, and speech
frames, in order as defined by their corresponding ToC entries in
the ToC list, contiguously into octets beginning with the most
significant bits of the fields and the octets.
To be precise, the four-bit payload header is packed into the first
octet of the payload with bit 0 of the payload header in the most
significant bit of the octet. The four most significant bits
(numbered 0-3) of the first ToC entry are packed into the least
significant bits of the octet, ending with bit 3 in the least
significant bit. Packing continues in the second octet with bit 4
of the first ToC entry in the most significant bit of the octet. If
more than one frame is contained in the payload, then packing
continues with the second and successive ToC entries. Bit 0 of the
first data frame follows immediately after the last ToC bit,
proceeding through all the bits of the frame in numerical order.
Bits from any successive frames follow contiguously in numerical
order for each frame and in consecutive order of the frames.
If speech data is missing for one or more speech frame within the
sequence, because of, for example, DTX, a ToC entry with FT set to
NO_DATA SHALL be included in the ToC for each of the missing frames,
but no data bits are included in the payload for the missing frame
(see Section 4.3.5.2 for an example).
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4.3.5. Payload Examples
4.3.5.1. Single Channel Payload Carrying a Single Frame
The following diagram shows a bandwidth-efficient AMR payload from a
single channel session carrying a single speech frame-block.
In the payload, no specific mode is requested (CMR=15), the speech
frame is not damaged at the IP origin (Q=1), and the coding mode is
AMR 7.4 kbps (FT=4). The encoded speech bits, d(0) to d(147), are
arranged in descending sensitivity order according to [2]. Finally,
two zero bits are added to the end as padding to make the payload
octet aligned.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CMR=15|0| FT=4 |1|d(0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| d(147)|P|P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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4.3.5.2. Single Channel Payload Carrying Multiple Frames
The following diagram shows a single channel, bandwidth efficient
compound AMR-WB payload that contains four frames, of which one has
no speech data. The first frame is a speech frame at 6.6 kbps mode
(FT=0) that is composed of speech bits d(0) to d(131). The second
frame is an AMR-WB SID frame (FT=9), consisting of bits g(0) to
g(39). The third frame is NO_DATA frame and does not carry any
speech information, it is represented in the payload by its ToC
entry. The fourth frame in the payload is a speech frame at 8.85
kbps mode (FT=1), it consists of speech bits h(0) to h(176).
As shown below, the payload carries a mode request for the encoder
on the receiver's side to change its future coding mode to AMR-WB
8.85 kbps (CMR=1). None of the frames are damaged at IP origin
(Q=1). The encoded speech and SID bits, d(0) to d(131), g(0) to
g(39) and h(0) to h(176), are arranged in the payload in descending
sensitivity order according to [4]. (Note, no speech bits are
present for the third frame). Finally, seven zero bits are padded
to the end to make the payload octet aligned.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CMR=1 |1| FT=0 |1|1| FT=9 |1|1| FT=15 |1|0| FT=1 |1|d(0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| d(131)|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|g(0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| g(39)|h(0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| h(176)|P|P|P|P|P|P|P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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4.3.5.3. Multi-Channel Payload Carrying Multiple Frames
The following diagram shows a two channel payload carrying 3
frame-blocks, i.e., the payload will contain 6 speech frames.
In the payload all speech frames contain the same mode 7.4 kbit/s
(FT=4) and are not damaged at IP origin. The CMR is set to 15,
i.e., no specific mode is requested. The two channels are defined
as left (L) and right (R) in that order. The encoded speech bits is
designated dXY(0).. dXY(K-1), where X = block number, Y = channel,
and K is the number of speech bits for that mode. Exemplifying
this, for frame-block 1 of the left channel the encoded bits are
designated as d1L(0) to d1L(147).
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CMR=15|1|1L FT=4|1|1|1R FT=4|1|1|2L FT=4|1|1|2R FT=4|1|1|3L FT|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|4|1|0|3R FT=4|1|d1L(0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| d1L(147)|d1R(0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| d1R(147)|d2L(0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|d2L(147|d2R(0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| d2R(147)|d3L(0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| d3L(147)|d3R(0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| d3R(147)|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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4.4. Octet-aligned Mode
4.4.1. The Payload Header
In octet-aligned mode, the payload header consists of a 4 bit CMR, 4
reserved bits, and optionally, an 8 bit interleaving header, as
shown below:
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+- - - - - - - -
| CMR |R|R|R|R| ILL | ILP |
+-+-+-+-+-+-+-+-+- - - - - - - -
CMR (4 bits): same as defined in section 4.3.1.
R: is a reserved bit that MUST be set to zero. All R bits MUST be
ignored by the receiver.
ILL (4 bits, unsigned integer): This is an OPTIONAL field that is
present only if interleaving is signalled out-of-band for the
session. ILL=L indicates to the receiver that the interleaving
length is L+1, in number of frame-blocks.
ILP (4 bits, unsigned integer): This is an OPTIONAL field that is
present only if interleaving is signalled. ILP MUST take a value
between 0 and ILL, inclusive, indicating the interleaving index
for frame-blocks in this payload in the interleave group. If the
value of ILP is found greater than ILL, the payload SHOULD be
discarded.
ILL and ILP fields MUST be present in each packet in a session if
interleaving is signalled for the session. Interleaving MUST be
performed on a frame-block basis (i.e., NOT on a frame basis) in a
multi-channel session.
The following example illustrates the arrangement of speech
frame-blocks in an interleave group during an interleave session.
Here we assume ILL=L for the interleave group that starts at speech
frame-block n. We also assume that the first payload packet of the
interleave group is s and the number of speech frame-blocks carried
in each payload is N. Then we will have:
Payload s (the first packet of this interleave group):
ILL=L, ILP=0,
Carry frame-blocks: n, n+(L+1), n+2*(L+1), ..., n+(N-1)*(L+1)
Payload s+1 (the second packet of this interleave group):
ILL=L, ILP=1,
frame-blocks: n+1, n+1+(L+1), n+1+2*(L+1), ..., n+1+(N-1)*(L+1)
...
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Payload s+L (the last packet of this interleave group):
ILL=L, ILP=L,
frame-blocks: n+L, n+L+(L+1), n+L+2*(L+1), ..., n+L+(N-1)*(L+1)
The next interleave group will start at frame-block n+N*(L+1).
There will be no interleaving effect unless the number of
frame-blocks per packet (N) is at least 2. Moreover, the number of
frame-blocks per payload (N) and the value of ILL MUST NOT be
changed inside an interleave group. In other words, all payloads in
an interleave group MUST have the same ILL and MUST contain the same
number of speech frame-blocks.
The sender of the payload MUST only apply interleaving if the
receiver has signalled its use through out-of-band means. Since
interleaving will increase buffering requirements at the receiver,
the receiver uses media type parameter "interleaving=I" to set the
maximum number of frame-blocks allowed in an interleaving group to
I.
When performing interleaving the sender MUST use a proper number of
frame-blocks per payload (N) and ILL so that the resulting size of
an interleave group is less or equal to I, i.e., N*(L+1)<=I.
4.4.2. The Payload Table of Contents and Frame CRCs
The table of contents (ToC) in octet-aligned mode consists of a list
of ToC entries where each entry corresponds to a speech frame
carried in the payload and, optionally, a list of speech frame CRCs,
i.e.,
+---------------------+
| list of ToC entries |
+---------------------+
| list of frame CRCs | (optional)
- - - - - - - - - - -
Note, for ToC entries with FT=14 or 15, there will be no
corresponding speech frame or frame CRC present in the payload.
The list of ToC entries is organized in the same way as described
for bandwidth-efficient mode in 4.3.2, with the following exception;
when interleaving is used the frame-blocks in the ToC will almost
never be placed consecutive in time. Instead, the presence and
order of the frame-blocks in a packet will follow the pattern
described in 4.4.1.
The following example shows the ToC of three consecutive packets,
each carrying 3 frame-blocks, in an interleaved two-channel session.
Here, the two channels are left (L) and right (R) with L coming
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before R, and the interleaving length is 3 (i.e., ILL=2). This
makes the interleave group 9 frame-blocks large.
Packet #1
---------
ILL=2, ILP=0:
+----+----+----+----+----+----+
| 1L | 1R | 4L | 4R | 7L | 7R |
+----+----+----+----+----+----+
|<------->|<------->|<------->|
Frame- Frame- Frame-
Block 1 Block 4 Block 7
Packet #2
---------
ILL=2, ILP=1:
+----+----+----+----+----+----+
| 2L | 2R | 5L | 5R | 8L | 8R |
+----+----+----+----+----+----+
|<------->|<------->|<------->|
Frame- Frame- Frame-
Block 2 Block 5 Block 8
Packet #3
---------
ILL=2, ILP=2:
+----+----+----+----+----+----+
| 3L | 3R | 6L | 6R | 9L | 9R |
+----+----+----+----+----+----+
|<------->|<------->|<------->|
Frame- Frame- Frame-
Block 3 Block 6 Block 9
A ToC entry takes the following format in octet-aligned mode:
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|F| FT |Q|P|P|
+-+-+-+-+-+-+-+-+
F (1 bit): see definition in Section 4.3.2.
FT (4 bits unsigned integer): see definition in Section 4.3.2.
Q (1 bit): see definition in Section 4.3.2.
P bits: padding bits, MUST be set to zero, and MUST be ignored on
reception.
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The list of CRCs is OPTIONAL. It only exists if the use of CRC is
signalled out-of-band for the session. When present, each CRC in
the list is 8 bit long and corresponds to a speech frame (NOT a
frame-block) carried in the payload. Calculation and use of the CRC
is specified in the next section.
4.4.2.1. Use of Frame CRC for UED over IP
The general concept of UED/UEP over IP is discussed in Section 3.6.
This section provides more details on how to use the frame CRC in
the octet-aligned payload header together with a partial transport
layer checksum to achieve UED.
To achieve UED, one SHOULD use a transport layer checksum, for
example, the one defined in UDP-Lite [19], to protect the IP,
transport protocol (e.g. UDP-Lite), and RTP headers, and in the
payload the payload header and the table of contents. The frame
CRC, when used, MUST be calculated only over all class A bits in the
AMR or AMR-WB frame. Class B and C bits in the AMR or AMR-WB frame
MUST NOT be included in the CRC calculation and SHOULD NOT be
covered by the transport checksum.
Note, the number of class A bits for various coding modes in AMR
codec is specified as informative in [2] and is therefore copied
into Table 1 in Section 3.6 to make it normative for this payload
format. The number of class A bits for various coding modes in
AMR-WB codec is specified as normative in table 2 in [4], and the
SID frame (FT=9) has 40 class A bits. These definitions of class
A bits MUST be used for this payload format.
If the transport layer checksum or link layer checksum detects any
errors within the protected (sensitive) part it is assumed that the
complete packet will be discarded as defined by UDP-Lite [19].
The receiver of the payload SHOULD examine the data integrity of the
received class A bits by re-calculating the CRC over the received
class A bits and comparing the result to the value found in the
received payload header. If the two values mismatch, the receiver
SHALL consider the class A bits in the receiver frame damaged and
MUST clear the Q flag of the frame (i.e., set it to 0). This will
subsequently cause the frame to be marked as SPEECH_BAD, if the FT
of the frame is 0..7 for AMR or 0..8 for AMR-WB, or SID_BAD if the
FT of the frame is 8 for AMR or 9 for AMR-WB, before it is passed to
the speech decoder. See [6] and [7] more details.
The following example shows an octet-aligned ToC with a CRC list for
a payload containing 3 speech frames from a single channel session
(assuming none of the FTs is equal to 14 or 15):
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| FT#1 |Q|P|P|1| FT#2 |Q|P|P|0| FT#3 |Q|P|P| CRC#1 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CRC#2 | CRC#3 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Each of the CRC's takes 8 bits
0 1 2 3 4 5 6 7
+---+---+---+---+---+---+---+---+
| c0| c1| c2| c3| c4| c5| c6| c7|
+---+---+---+---+---+---+---+---+
(MSB) (LSB)
and is calculated by the cyclic generator polynomial,
C(x) = 1 + x^2 + x^3 + x^4 + x^8
where ^ is the exponentiation operator.
In binary form the polynomial has the following form: 101110001
(MSB..LSB).
The actual calculation of the CRC is made as follows: First, an
8-bit CRC register is reset to zero: 00000000. For each bit over
which the CRC shall be calculated, an XOR operation is made between
the rightmost (LSB) bit of the CRC register and the bit. The CRC
register is then right shifted one step (each bit's significance is
reduced by one), inputting a "0" as the leftmost bit (MSB). If the
result of the XOR operation mentioned above is a "1" then "10111000"
is bit-wise XOR-ed into the CRC register. This operation is
repeated for each bit that the CRC should cover. In this case, the
first bit would be d(0) for the speech frame for which the CRC
should cover. When the last bit (e.g., d(54) for AMR 5.9 according
to Table 1 in Section 3.6) have been used in this CRC calculation,
the contents in CRC register should simply be copied to the
corresponding field in the list of CRC's.
Fast calculation of the CRC on a general-purpose CPU is possible
using a table-driven algorithm.
4.4.3. Speech Data
In octet-aligned mode, speech data is carried in a similar way to
that in the bandwidth-efficient mode as discussed in Section 4.3.3,
with the following exceptions:
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- The last octet of each speech frame MUST be padded with zero
bits at the end if not all bits in the octet are used. The
padding bits MUST be ignored on reception. In other words,
each speech frame MUST be octet-aligned.
- When multiple speech frames are present in the speech data
(i.e., compound payload), the speech frames can be arranged
either one whole frame after another as usual, or with the
octets of all frames interleaved together at the octet level.
Since the bits within each frame are ordered with the most
error-sensitive bits first, interleaving the octets collects
those sensitive bits from all frames to be nearer the
beginning of the packet. This is called "robust sorting
order" which allows the application of UED (such as UDP-Lite
[19]) or UEP (such as the ULP [22]) mechanisms to the payload
data. The details of assembling the payload are given in the
next section.
The use of robust sorting order for a session MUST be agreed via
out-of-band means. Section 8 specifies a media type parameter for
this purpose.
Note, robust sorting order MUST only be performed on the frame level
and thus is independent of interleaving which is at the frame-block
level, as described in Section 4.4.1. In other words, robust sorting
can be applied to either non-interleaved or interleaved sessions.
4.4.4. Methods for Forming the Payload
Two different packetization methods, namely normal order and robust
sorting order, exist for forming a payload in octet-aligned mode.
In both cases, the payload header and table of contents are packed
into the payload the same way; the difference is in the packing of
the speech frames.
The payload begins with the payload header of one octet or two if
frame interleaving is selected. The payload header is followed by
the table of contents consisting of a list of one-octet ToC entries.
If frame CRCs are to be included, they follow the table of contents
with one 8-bit CRC filling each octet. Note that if a given frame
has a ToC entry with FT=14 or 15, there will be no CRC present.
The speech data follows the table of contents, or the CRCs if
present. For packetization in the normal order, all of the octets
comprising a speech frame are appended to the payload as a unit. The
speech frames are packed in the same order as their corresponding
ToC entries are arranged in the ToC list, with the exception that if
a given frame has a ToC entry with FT=14 or 15, there will be no
data octets present for that frame.
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For packetization in robust sorting order, the octets of all speech
frames are interleaved together at the octet level. That is, the
data portion of the payload begins with the first octet of the first
frame, followed by the first octet of the second frame, then the
first octet of the third frame, and so on. After the first octet of
the last frame has been appended, the cycle repeats with the second
octet of each frame. The process continues for as many octets as
are present in the longest frame. If the frames are not all the
same octet length, a shorter frame is skipped once all octets in it
have been appended. The order of the frames in the cycle will be
sequential if frame interleaving is not in use, or according to the
interleave pattern specified in the payload header if frame
interleaving is in use. Note that if a given frame has a ToC entry
with FT=14 or 15, there will be no data octets present for that
frame so that frame is skipped in the robust sorting cycle.
The UED and/or UEP is RECOMMENDED to cover at least the RTP header,
payload header, table of contents, and class A bits of a sorted
payload. Exactly how many octets need to be covered depends on the
network and application. If CRCs are used together with robust
sorting, only the RTP header, the payload header, and the ToC SHOULD
be covered by UED/UEP. The means to communicate to other layers
performing UED/UEP the number of octets to be covered is beyond the
scope of this specification.
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4.4.5. Payload Examples
4.4.5.1. Basic Single Channel Payload Carrying Multiple Frames
The following diagram shows an octet aligned payload from a single
channel session that carries two AMR frames of 7.95 kbps coding mode
(FT=5). In the payload, a codec mode request is sent (CMR=6),
requesting the encoder at the receiver's side to use AMR 10.2 kbps
coding mode. No frame CRC, interleaving, or robust-sorting is in
use.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CMR=6 |R|R|R|R|1|FT#1=5 |Q|P|P|0|FT#2=5 |Q|P|P| f1(0..7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| f1(8..15) | f1(16..23) | .... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... |f1(152..158) |P| f2(0..7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| f2(8..15) | f2(16..23) | .... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... |f2(152..158) |P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Note, in above example the last octet in both speech frames is
padded with one zero bit to make it octet-aligned.
4.4.5.2. Two Channel Payload with CRC, Interleaving, and Robust-sorting
This example shows an octet aligned payload from a two channel
session. Two frame-blocks, each containing two speech frames of
7.95 kbps coding mode (FT=5), are carried in this payload,
The two channels are left (L) and right (R) with L coming before R.
In the payload, a codec mode request is also sent (CMR=6),
requesting the encoder at the receiver's side to use AMR 10.2 kbps
coding mode.
Moreover, frame CRC and frame-block interleaving are both enabled
for the session. The interleaving length is 2 (ILL=1) and this
payload is the first one in an interleave group (ILP=0).
The first two frames in the payload are the L and R channel speech
frames of frame-block #1, consisting of bits f1L(0..158) and
f1R(0..158), respectively. The next two frames are the L and R
channel frames of frame-block #3, consisting of bits f3L(0..158) and
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f3R(0..158), respectively, due to interleaving. For each of the
four speech frames a CRC is calculated as CRC1L(0..7), CRC1R(0..7),
CRC3L(0..7), and CRC3R(0..7), respectively. Finally, the payload is
robust sorted.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CMR=6 |R|R|R|R| ILL=1 | ILP=0 |1|FT#1L=5|Q|P|P|1|FT#1R=5|Q|P|P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1|FT#3L=5|Q|P|P|0|FT#3R=5|Q|P|P| CRC1L | CRC1R |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CRC3L | CRC3R | f1L(0..7) | f1R(0..7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| f3L(0..7) | f3R(0..7) | f1L(8..15) | f1R(8..15) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| f3L(8..15) | f3R(8..15) | f1L(16..23) | f1R(16..23) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| f3L(144..151) | f3R(144..151) |f1L(152..158)|P|f1R(152..158)|P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|f3L(152..158)|P|f3R(152..158)|P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Note, in above example the last octet in all the four speech frames
is padded with one zero bit to make it octet-aligned.
4.5. Implementation Considerations
An application implementing this payload format MUST understand all
the payload parameters in the out-of-band signaling used. For
example, if an application uses SDP, all the SDP and media type
parameters in this document MUST be understood. This requirement
ensures that an implementation always can decide if it is capable or
not of communicating.
No operating mode of the payload format is mandatory to implement.
The requirements of the application using the payload format should
be used to determine what to implement. To achieve basic
interoperability an implementation SHOULD at least implement both
bandwidth-efficient and octet-aligned modes for a single audio
channel. The other operating modes: interleaving, robust sorting,
and frame-wise CRC in both single and multi-channel, are OPTIONAL to
implement.
The mode-change period and mode-change-neighbor parameters are
intended for signaling with GSM endpoints. When interoperability
with GSM is desired, encoders SHOULD only perform codec mode changes
to neighboring modes and in integer multiples of 40ms (two
frame-blocks), but decoders SHOULD accept codec mode changes at any
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time, i.e. for every frame-block. The encoder may arbitrarily select
the initial phase (odd or even frame-block), where codec mode
changes are performed, but then SHOULD stick to that phase as far as
possible. Handovers or other events (e.g. call forwarding) may,
however, in rare cases change this phase and may also cause mode
changes to non-neighboring modes. The decoder SHALL therefore be
prepared to accept changes also in the other phase and to other
modes. Section 8 specifies the usage of the parameters mode-change-
period and mode-change-capability to indicate the desired behavior
in applications.
See 3GPP TS 26.103 [28] for preferred AMR and AMR-WB configurations
for operation in GSM and 3GPP UMTS networks. In gateway scenarios
encoders can be requested through the "mode-set" parameter to use a
limited mode-set that is supported by the link beyond the gateway.
Further to avoid congestion on that link, the encoder SHOULD limit
the initial codec mode for a session to a lower mode, until at least
one frame-block is received with rate control information.
4.5.1. Decoding Validation
When processing a received payload packet, if the receiver finds
that the calculated payload length, based on the information of the
session and the values found in the payload header fields, does not
match the size of the received packet, the receiver SHOULD discard
the packet. This is because decoding a packet that has errors in
its length field could severely degrade the speech quality.
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5. AMR and AMR-WB Storage Format
The storage format is used for storing AMR or AMR-WB speech frames
in a file or as an e-mail attachment. Multiple channel content is
supported.
In general, an AMR or AMR-WB file has the following structure:
+------------------+
| Header |
+------------------+
| Speech frame 1 |
+------------------+
: ... :
+------------------+
| Speech frame n |
+------------------+
Note, to preserve interoperability with already deployed
implementations, single channel content uses a file header format
different from that of multi-channel content.
There also exists another storage format for AMR and AMR-WB that is
suitable for applications with more advanced demands on the storage
format, like random access or synchronization with video. This
format is the 3GPP specified ISO-based multi-media file format 3GP
[31]. Its media type is specified by RFC 3839 [32].
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5.1. Single channel Header
A single channel AMR or AMR-WB file header contains only a magic
number and different magic numbers are defined to distinguish AMR
from AMR-WB.
The magic number for single channel AMR files MUST consist of ASCII
character string:
"#!AMR\n"
(or 0x2321414d520a in hexadecimal).
The magic number for single channel AMR-WB files MUST consist of
ASCII character string:
"#!AMR-WB\n"
(or 0x2321414d522d57420a in hexadecimal).
Note, the "\n" is an important part of the magic numbers and MUST be
included in the comparison, since, otherwise, the single channel
magic numbers above will become indistinguishable from those of the
multi-channel files defined in the next section.
5.2. Multi-channel Header
The multi-channel header consists of a magic number followed by a
32-bit channel description field, giving the multi-channel header
the following structure:
+------------------+
| magic number |
+------------------+
| chan-desc field |
+------------------+
The magic number for multi-channel AMR files MUST consist of the
ASCII character string:
"#!AMR_MC1.0\n"
(or 0x2321414d525F4D43312E300a in hexadecimal).
The magic number for multi-channel AMR-WB files MUST consist of the
ASCII character string:
"#!AMR-WB_MC1.0\n"
(or 0x2321414d522d57425F4D43312E300a in hexadecimal).
The version number in the magic numbers refers to the version of the
file format.
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The 32 bit channel description field is defined as:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Reserved bits | CHAN |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Reserved bits: MUST be set to 0 when written, and a reader MUST
ignore them.
CHAN (4 bit unsigned integer): Indicates the number of audio
channels contained in this storage file. The valid values and the
order of the channels within a frame block are specified in Section
4.1 in [12].
5.3. Speech Frames
After the file header, speech frame-blocks consecutive in time are
stored in the file. Each frame-block contains a number of
octet-aligned speech frames equal to the number of channels, and
stored in increasing order, starting with channel 1.
Each stored speech frame starts with a one octet frame header with
the following format:
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|P| FT |Q|P|P|
+-+-+-+-+-+-+-+-+
The FT field and the Q bit are defined in the same way as in Section
4.3.2 The P bits are padding and MUST be set to 0, and MUST be
ignored.
Following this one octet header come the speech bits as defined in
4.4.3 The last octet of each frame is padded with zeroes, if
needed, to achieve octet alignment.
The following example shows an AMR frame in 5.9 kbit coding mode
(with 118 speech bits) in the storage format.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|P| FT=2 |Q|P|P| |
+-+-+-+-+-+-+-+-+ +
| |
+ Speech bits for frame-block n, channel k +
| |
+ +-+-+
| |P|P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Non-received speech frames or frame-blocks between SID updates
during non-speech periods MUST be stored as NO_DATA frames (frame
type 15, as defined in [2] and [4]). Frames or frame-blocks lost in
transmission MUST be stored as NO_DATA frames or SPEECH_LOST (frame
type 14, only available for AMR-WB) in complete frame-blocks to keep
synchronization with the original media.
Comfort noise frames of other types than AMR SID (FT=8), i.e. frame
type 9,10 and 11 for AMR, SHALL NOT be used in the AMR file format.
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].
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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
signalling protocol employed. Therefore a single mechanism is not
sufficient, although if suitable the usage of SRTP [26] is
RECOMMENDED. Other known mechanism that may be used are IPsec [33]
and TLS [34] (RTP over TCP), but also other alternatives may 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 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 end points. 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 interleave groups. If it is
not done at that boundary on both endpoints, the speech quality will
be degraded during the complete interleave 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
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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 speech in a different 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 3555 [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 e-mail 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
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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 session.
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 two
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 of
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
client has the capability to restrict the mode change
period to 2, 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 (value = 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.
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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 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 RFC2327 [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 elapse 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.
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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 XXXX.
Public specification:
Please refer to Section 11 of RFC XXXX.
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)
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].
File extensions: amr, AMR
Macintosh file type code: "amr " (fourth character is space)
Person & email address to contact for further information:
magnus.westerlund@ericsson.com
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 SHALL the RTP payload format specified in
Section 4 be used. In all other context SHALL the file
format defined in Section 5 be used.
Author:
magnus.westerlund@ericsson.com
ari.lakaniemi@nokia.com
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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 session.
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 two
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.
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mode-change-capability: Specifies if the client is capable of
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
client has the capability to restrict the mode change
period to 2, 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 (value = 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 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
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used. The presence of this parameter also implies
automatically that octet-aligned operation SHALL be
used.
ptime: see RFC2327 [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 elapse 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 XXXX.
Public specification:
Please refer to Section 11 of RFC XXXX.
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].
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Person & email address to contact for further information:
magnus.westerlund@ericsson.com
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 SHALL the RTP payload format specified in
Section 4 be used. In all other context SHALL the file
format defined in Section 5 be used.
Author:
magnus.westerlund@ericsson.com
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
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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-align
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 to not 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 is
primarily intended for gateways to networks such as GSM or
3GPP UMTS, which transport only supports a subset. The 3GPP
preferred codec configurations are defined in 3GPP TS
26.103 [25], and it is RECOMMENDED that also other networks
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, only if no mode-set was supplied
in the offer for a unicast two-peer session, is the
answerer free to choose a mode-set in the answer. 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.
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- 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 the offerer 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 answer's capabilty 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.
- The parameters "maxptime" and "ptime" will in most cases
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 RFC3264 [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
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answerer MAY change the value but is RECOMMEDED 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 is
RECOMMEDED to always include the parameter and bound its
usage of redundancy to simplify for 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 orignally 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.
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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.
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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) are updated, see
Section 8.
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10. Changes
The differences between RFC 3267 and this document are as follows:
- Added clarification what 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 negoatiation of the payload
parameters.
- Clarified and aligned the unequal detection usage with the
published UDP-Lite specification in section 3.6.1 and 4.4.2.1.
This including removing 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 Section 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
3711, RFC 3828, RFC 3550, RFC 3448, and RFC 3551. 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 may 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
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addition of this parameter should have no negative effects on
older implementations as they are mandated to ignore unknown
parameters per RFC 3267. And in addition 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 consideration
to Implementation consideration 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 update. The authors would also like
to thank Richard Ejzak, Thomas Belling, and Gorry Fairhurst for
their input on this update of RFC 3267.
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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., V. Jacobson 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",
draft-ietf-avt-rfc3555bis-04, April 17, 2006.
Sjoberg, et. al. Standards Track [Page 55]
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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.
Sjoberg, et. al. Standards Track [Page 56]
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ETSI documents can be downloaded from the ETSI web server,
http://www.etsi.org/". Any 3GPP document can be downloaded from the
3GPP webserver, "http://www.3gpp.org/", see specifications. TIA
documents can be obtained from "www.tiaonline.org".
13. 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: qxie1@email.mot.com
Sjoberg, et. al. Standards Track [Page 57]
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14. IPR Notice
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Copyright (C) The Internet Society (2006).
This document is subject to the rights, licenses and restrictions
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This document and the information contained herein are provided on
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This Internet-Draft expires in January 2007
RFC Editor Considerations
- The RFC editor is requested to replace all occurances of XXXX
with the RFC number this document receives.
Sjoberg, et. al. Standards Track [Page 58]
