Network Working Group Johan Sjoberg
INTERNET-DRAFT Magnus Westerlund
Expires: March 2006 Ericsson
Ari Lakaniemi
Stephan Wenger
Nokia
September 22, 2005
RTP Payload Format for Extended AMR Wideband (AMR-WB+) Audio Codec
<draft-ietf-avt-rtp-amrwbplus-07.txt>
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Abstract
This document specifies a real-time transport protocol (RTP) payload
format for Extended Adaptive Multi-Rate Wideband (AMR-WB+) encoded
audio signals. The AMR-WB+ codec is an audio extension of the AMR-
WB speech codec. It encompasses the AMR-WB frame types and a number
of new frame types designed to support high quality music and
speech. A media type registration for AMR-WB+ is included in this
specification.
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TABLE OF CONTENTS
1. Definitions.....................................................3
1.1. Glossary...................................................3
1.2. Terminology................................................3
2. Introduction....................................................3
3. Background of AMR-WB+ and Design Principles.....................4
3.1. The AMR-WB+ Audio Codec....................................4
3.2. Multi-rate Encoding and Rate Adaptation....................7
3.3. Voice Activity Detection and Discontinuous Transmission....8
3.4. Support for Multi-Channel Session..........................8
3.5. Unequal Bit-error Detection and Protection.................8
3.6. Robustness against Packet Loss.............................9
3.6.1. Use of Forward Error Correction (FEC).................9
3.6.2. Use of Frame Interleaving............................10
3.7. AMR-WB+ Audio over IP scenarios...........................11
3.8. Out-of-Band Signaling.....................................12
4. RTP Payload Format for AMR-WB+.................................12
4.1. RTP Header Usage..........................................13
4.2. Payload Structure.........................................14
4.3. Payload Definitions.......................................14
4.3.1. Payload Header.......................................14
4.3.2. The Payload Table of Contents........................15
4.3.3. Audio Data...........................................21
4.3.4. Methods for Forming the Payload......................21
4.3.5. Payload Examples.....................................22
4.4. Interleaving Considerations...............................24
4.5. Implementation Considerations.............................25
4.5.1. ISF recovery in case of packet loss..................26
4.5.2. Decoding Validation..................................28
5. Congestion Control.............................................28
6. Security Considerations........................................28
6.1. Confidentiality...........................................29
6.2. Authentication and Integrity..............................29
7. Payload Format Parameters......................................29
7.1. Media Type Registration...................................30
7.2. Mapping Media Type Parameters into SDP....................31
7.2.1. Offer-Answer Model Considerations....................32
7.2.2. Examples.............................................34
8. IANA Considerations............................................34
9. Contributors...................................................34
10. Acknowledgements..............................................34
11. References....................................................35
11.1. Normative references.....................................35
11.2. Informative references...................................36
12. Authors' Addresses............................................37
13. IPR Notice....................................................38
14. Copyright Notice..............................................38
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1. Definitions
1.1. Glossary
3GPP - Third Generation Partnership Project
AMR - Adaptive Multi-Rate (Codec)
AMR-WB - Adaptive Multi-Rate Wideband (Codec)
AMR-WB+ - Extended Adaptive Multi-Rate Wideband (Codec)
CMR - Codec Mode Request
CN - Comfort Noise
DTX - Discontinuous Transmission
FEC - Forward Error Correction
FT - Frame Type
ISF - Internal Sampling Frequency
SCR - Source Controlled Rate Operation
SID - Silence Indicator (the frames containing only CN
parameters)
TFI - Transport Frame Index
TS - Timestamp
VAD - Voice Activity Detection
UED - Unequal Error Detection
UEP - Unequal Error Protection
1.2. Terminology
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 RFC 2119 [2].
2. Introduction
This document specifies the payload format for packetization of
Extended Adaptive Multi-Rate Wideband (AMR-WB+) [1] encoded audio
signals into the Real-time Transport Protocol (RTP) [3]. The
payload format supports the transmission of mono or stereo audio,
aggregating multiple frames per payload, and mechanisms enhancing
the robustness of the packet stream against packet loss.
The AMR-WB+ codec is an extension of the Adaptive Multi-Rate
Wideband (AMR-WB) speech codec. New features include extended audio
bandwidth to enable high quality for non-speech signals (e.g.
music), native support for stereophonic audio, and the option to
operate on, and switch between, several internal sampling
frequencies (ISFs). The primary usage scenario for AMR-WB+ is the
transport over IP. Therefore, interworking with other transport
networks, as discussed for AMR-WB in [7], is not a major concern and
hence not addressed in this memo.
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The expected key application for AMR-WB+ is streaming. To make the
packetization process on a streaming server as efficient as
possible, an octet-aligned payload format is desirable. Therefore,
a bandwidth efficient mode as defined for AMR-WB in [7] is not
specified herein; the bandwidth-savings of the bandwidth efficient
mode would be very small anyway, since all extension frame types are
octet aligned.
The stereo encoding capability of AMR-WB+ renders the support for
multi-channel transport at RTP payload format level, as specified
for AMR-WB [7], obsolete. Therefore this feature is not included in
this memo.
This specification does not include a definition of a file format
for AMR-WB+. Instead, it is referred to the ISO based 3GP file
format [14], which supports AMR-WB+ and provides all functionality
required. The 3GP format also supports storage of AMR and AMR-WB,
and many other multi-media formats, thereby allowing synchronized
playback.
The rest of the document is organized as follows: Background
information on the AMR-WB+ codec, and design principles, can be
found in Section 3. The payload format itself is specified in
Section 4. Sections 5 and 6 discuss congestion control and security
considerations, respectively. In Section 7, a media type
registration is provided.
3. Background of AMR-WB+ and Design Principles
The Extended Adaptive Multi-Rate Wideband (AMR-WB+) [1] audio codec
is designed to compress speech and audio signals at low bit-rate and
good quality. The codec is specified by the Third Generation
Partnership Project (3GPP). The primary target applications are 1.
the packet-switched streaming service (PSS) [13], 2. multimedia
messaging service (MMS), and 3. multimedia broadcast and multicast
service (MBMS). However, due to its flexibility and robustness, AMR-
WB+ is also well suited for streaming services in other highly
varying transport environments, for example the Internet.
3.1. The AMR-WB+ Audio Codec
3GPP originally developed the AMR-WB+ audio codec for streaming and
messaging services in Global System for Mobile communications (GSM)
and third generation (3G) cellular systems. The codec is designed
as an audio extension of the AMR-WB speech codec. The extension
adds new functionality to the codec in order to provide high audio
quality for a large range of signals including music. Stereophonic
operation has also been added. A new, high-efficiency hybrid stereo
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coding algorithm enables stereo operation at bit-rates as low as 6.2
kbit/s.
The AMR-WB+ codec includes the nine frame types specified for AMR-
WB, extended by new bit-rates ranging from 5.2 to 48 kbit/s. The
AMR-WB frame types can employ only a 16000 Hz sampling frequency and
operate only on monophonic signals. The newly introduced extension
frame types, however, can operate at a number of internal sampling
frequencies (ISFs), both in mono and stereo. Please see Table 24 in
[1] for details. The output sampling frequency of the decoder is
limited to 8, 16, 24, 32 or 48 kHz.
An overview of the AMR-WB+ encoding operations is provided as
follows. The encoder receives the audio sampled at, for example, 48
kHz. The encoding process starts with pre-processing and resampling
to the user-selected ISF. The encoding is performed on equally
sized super-frames. Each super-frame corresponds to 2048 samples
per channel, at the ISF. The codec carries out a number of encoding
decisions for each super-frame, thereby choosing between different
encoding algorithms and block lengths, so to achieve a fidelity-
optimized encoding adapted to the signal characteristics of the
source. The stereo encoding (if used) executes separately from the
monophonic core encoding, thus enabling the selection of different
combinations of core and stereo encoding rates. The resulting
encoded audio is produced in four transport frames of equal length.
Each transport frame corresponds to 512 samples at the ISF, and is
individually usable by the decoder, provided that its position in
the super-frame structure is known.
The codec supports 13 different ISFs, ranging from 12.8 up to 38.4
kHz, as described by Table 24 of [1]. The high number of ISFs
allows a trade-off between the audio bandwidth and the target bit-
rate. As encoding is performed on 2048 samples at the ISF, the
duration of a super-frame and the effective bit-rate of the frame
type in use varies.
The ISF of 25600 Hz has a super-frame duration of 80 ms. It is the
'nominal' value used to describe the encoding bit-rates henceforth.
Assuming this normalization, the ISF selection results in bit-rate
variations from 1/2 up to 3/2 of the nominal bit-rate.
The encoding for the extension modes is performed as one monophonic
core encoding and one stereo encoding. The core encoding is
executed by splitting the monophonic signal into a lower and a
higher frequency band. The lower band is encoded employing either
algebraic code excited linear prediction (ACELP), or transform coded
excitation (TCX). This selection can be made once per transport
frame, but must obey certain limitations of legal combinations
within the super-frame. The higher band is encoded using a low-rate
parametric bandwidth extension approach.
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The stereo signal is encoded employing a similar frequency band
decomposition; however, here the signal is divided into three bands
that are individually parameterized.
The total bit-rate produced by the extension is the result of the
combination of the encoder's core rate, stereo rate and ISF. The
extension supports 8 different core encoding rates producing bit-
rates between 10.4 and 24.0 kbit/s; see table 22 in [1]. There are
16 stereo encoding rates generating bit-rates between 2.0 and 8.0
kbit/s; see table 23 in [1]. The frame type encodes the AMR-WB
modes, 4 fixed extension rates (see below), 24 combinations of core
and stereo rates for stereo signals, and the 8 core rates for mono
signals, as listed in table 25 in [1]. This results in the AMR-WB+
supporting encoding rates between 10.4 and 32 kbit/s, assuming an
ISF of 25600 Hz.
Different ISFs allow for additional freedom in the produced bit-
rates and audio quality. The selection of an ISF changes the
available audio bandwidth of the reconstructed signal, and also the
total bit-rate. The bit-rate for a given combination of frame type
and ISF is determined by multiplying the frame type's bit-rate with
the used ISF's bit-rate factor, see table 24 in [1].
The extension also has four frame types which have fixed ISFs.
Please see frame types 10-13 in Table 21 in [1]. These four pre-
defined frame types have a fixed input sampling frequency at the
encoder, which can be set either at 16 or 24 kHz. Like the AMR-WB
frame types, transport frames encoded utilizing these frame types
represent exactly 20 ms of the audio signal. However, they are also
part of 80 ms super-frames. Frame types 0-13 (AMR-WB and fixed
extension rates), as listed in table 21 in [1], do not require an
explicit ISF indication. The other frame types 14-47 require the
ISF employed to be indicated.
The 32 different frame types of the extension, in combination with
13 ISFs, allows for a great flexibility in bit-rate and selection of
desired audio quality. A number of combinations exist that produce
the same codec bit-rate. For example, a 32 kbit/s audio stream can
be produced by utilizing frame type 41, i.e. 25.6 kbit/s, and the
ISF of 32kHz (5/4 * (19.2+6.4) = 32 kbit/s), or frame type 47 and
the ISF of 25.6 kHz (1 * (24 + 8) = 32 kbit/s). Which combination
is more beneficial for the perceived audio quality depends on the
content. In the above example the first case provides a higher
audio bandwidth, while the second one spends the same number of bits
on somewhat narrower audio bandwidth but provides higher fidelity.
Encoders are free to select the combination they deem most
beneficial.
Since a transport frame always corresponds to 512 samples at the
used ISF, its duration is limited to the range 13.33 to 40 ms, see
Table 1. An RTP Timestamp clock rate of 72000 Hz, as mandated by
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this specification, results in AMR-WB+ transport frame lengths of
960 to 2880 timestamp ticks, depending solely on the selected ISF.
Index ISF Duration(ms) Duration(TS Ticks @ 72 kHz)
------------------------------------------------------
0 N/A 20 1440
1 12800 40 2880
2 14400 35.55 2560
3 16000 32 2304
4 17067 30 2160
5 19200 26.67 1920
6 21333 24 1728
7 24000 21.33 1536
8 25600 20 1440
9 28800 17.78 1280
10 32000 16 1152
11 34133 15 1080
12 36000 14.22 1024
13 38400 13.33 960
Table 1: Normative number of RTP Timestamp Ticks for each
Transport Frame depending on ISF (ISF and Duration in
ms are rounded)
The encoder is free to change both the ISF and the encoding frame
type (both mono and stereo) during a session. For the extension
frame types with index 10-13 and 16-47, the ISF and frame type
changes are constrained to occur at super-frame boundaries. This
implies that, for the frame types mentioned, the ISF is constant
throughout a super-frame. This limitation does not apply for frame
types with index 0-9, 14 and 15, i.e. the original AMR-WB frame
types.
A number of features of the AMR-WB+ codec require special
consideration from a transport point of view, and solutions that
could perhaps be viewed as unorthodox. First, there are constraints
on the RTP timestamping, due to the relationship of the frame
duration and the ISFs. Second, each frame of encoded audio must
maintain information about its frame type, ISF and position in the
super-frame.
3.2. Multi-rate Encoding and Rate Adaptation
The multi-rate encoding capability of AMR-WB+ is designed to
preserve high audio quality under a wide range of bandwidth
requirements and transmission conditions.
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AMR-WB+ enables seamless switching between frame types that use the
same number of audio channels and the same ISF. Every AMR-WB+ codec
implementation is required to support all frame types defined by the
codec, and must be able to handle switching between any two frame
types. Switching between frame types employing a different number
of audio channels or a different ISF must also be supported, but it
may not be completely seamless. Therefore it is recommended to
perform such switching infrequently and, if possible, during periods
of silence.
3.3. Voice Activity Detection and Discontinuous Transmission
AMR-WB+ supports the same algorithms as AMR-WB for voice activity
detection (VAD) and generation of comfort noise (CN) parameters
during silence periods. However, these functionalities can only be
used in conjunction with the AMR-WB frame types (FT=0-8). This
option allows reducing 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-WB+ frames containing CN parameters are
called Silence Indicator (SID) frames. More details about the VAD
and DTX functionality is provided in [4] and [5].
3.4. Support for Multi-Channel Session
Some of the AMR-WB+ frame types support the encoding of stereophonic
audio. Because of this native support for a two-channel
stereophonic signal, it does not seem necessary to support multi-
channel transport with separate codec instances, as specified in the
AMR-WB RTP payload [7]. The codec has the capability of stereo to
mono downmixing as part of the decoding process. Thus, a receiver
that is only capable of playout of monophonic audio must still be
able to decode and play signals originally encoded and transmitted
as stereo. However, to avoid spending bits on a stereo encoding
that is not going to be utilized, a mechanism is defined in this
specification to signal mono-only audio.
3.5. Unequal Bit-error Detection and Protection
The audio bits encoded in each AMR-WB frame are sorted according to
their different perceptual sensitivity to bit errors. In cellular
systems, for example, this property can be exploited to achieve
better voice quality, by using unequal error protection and
detection (UEP and UED) mechanisms. However, the bits of the
extension frame types of the AMR-WB+ codec do not have a consistent
perceptual significance property and are not sorted in this order.
Thus, UEP or UED is meaningless with the extension frame types. If
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there is a need to use UEP or UED for AMR-WB frame types, it is
recommended to use RFC 3267 [7].
3.6. Robustness against Packet Loss
The payload format supports two mechanisms to improve robustness
against packet loss: simple forward error correction (FEC) and frame
interleaving.
3.6.1. Use of Forward Error Correction (FEC)
Generic forward error correction within RTP is defined, for example
in RFC2733 [11]. Audio redundancy coding is defined in RFC2198
[12]. Either scheme can be used to add redundant information to the
RTP packet stream and make it more resilient to packet losses, at
the expense of a higher bit rate. Please see either RFC for a
discussion of the implications of the higher bit rate to network
congestion.
In addition to these media-unaware mechanisms, this memo specifies
an AMR-WB+ specific form of audio redundancy coding, which may be
beneficial in terms of packetization overhead.
Conceptually, previously transmitted transport frame(s) are
aggregated together with new one(s). A sliding window is used to
group the frames to be sent in each payload. Figure 1 below shows
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.
Here, each frame is retransmitted once in the following RTP payload
packet. F(n-2)...f(n+4) denote a sequence of audio frames and p(n-
1)...p(n+4) a sequence of payload packets.
The mechanism described does not require signaling at the session
setup. In other words, the audio sender can choose to use this
scheme without consulting the receiver. For a certain timestamp,
the receiver may receive multiple copies of a frame containing
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encoded audio data or frames indicated as NO_DATA. The cost of this
scheme is bandwidth and the receiver delay necessary to allow the
redundant copy to arrive.
This redundancy scheme provides a similar functionality as the one
described in RFC 2198, but works only if both original frames and
redundant representations are AMR-WB+ frames. When the use of other
media coding schemes is desirable, one has to resort to RFC2198.
The sender is responsible for selecting an appropriate amount of
redundancy based on feedback about the channel conditions, e.g. in
the RTP Control Protocol (RTCP) [3] receiver reports. The sender is
also responsible for avoiding congestion, which may be exacerbated
by redundancy (see Section 5 for more details).
3.6.2. Use of Frame Interleaving
To decrease protocol overhead, the payload design allows several
audio transport frames to be encapsulated into a single RTP packet.
One of the drawbacks of such an approach is that in case of packet
loss several consecutive frames are lost. Consecutive frame loss
normally renders error concealment less efficient and usually causes
clearly audible and annoying distortions in the reconstructed audio.
Interleaving of transport frames can improve the audio quality in
such cases by distributing the consecutive losses into a number of
isolated frame losses, which are easier to conceal. However,
interleaving and bundling several frames per payload also increases
end-to-end delay and sets higher buffering requirements. Therefore,
interleaving is not appropriate for all use cases or devices.
Streaming applications should most likely be able to exploit
interleaving to improve audio quality in lossy transmission
conditions.
Note that this payload design supports the use of frame interleaving
as an option. The usage of this feature needs to be negotiated in
the session set-up.
The interleaving supported by this format is rather flexible. For
example, a continuous pattern can be defined, as depicted in Figure
2.
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--+--------+--------+--------+--------+--------+--------+--------+--
| f(n-2) | f(n-1) | f(n) | f(n+1) | f(n+2) | f(n+3) | f(n+4) |
--+--------+--------+--------+--------+--------+--------+--------+--
[ P(n) ]
[ P(n+1) ] [ P(n+1) ]
[ P(n+2) ] [ P(n+2) ]
[ P(n+3) ] [P(
[ P(n+4) ]
Figure 2: An example of interleaving pattern that has constant
delay.
In Figure 2 the consecutive frames, denoted f(n-2) to f(n+4), are
aggregated into packets P(n) to P(n+4), each packet carrying two
frames. This approach provides an interleaving pattern that allows
for constant delay in both the interleaving and deinterleaving
processes. The deinterleaving buffer needs to have room for at
least three frames, including the one that is ready to be consumed.
The storage space for three frames is needed, for example, when f(n)
is the next frame to be decoded: since frame f(n) was received in
packet P(n+2) carrying also frame f(n+3), both these frames are
stored in the buffer. Furthermore, frame f(n+1) received in the
previous packet P(n+1) is also in the deinterleaving buffer. Note
also that in this example the buffer occupancy varies: when frame
f(n+1) is the next one to be decoded, there are only two frames,
f(n+1) and f(n+3), in the buffer.
3.7. AMR-WB+ Audio over IP scenarios
Since the primary target application for the AMR-WB+ codec is
streaming over packet networks, the most relevant usage scenario for
this payload format is IP end-to-end between a server and a
terminal, as shown in Figure 3.
+----------+ +----------+
| | IP/UDP/RTP/AMR-WB+ | |
| SERVER |<------------------------>| TERMINAL |
| | | |
+----------+ +----------+
Figure 3: Server to terminal IP scenario
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3.8. Out-of-Band Signaling
Some of the options of this payload format remain constant
throughout a session. Therefore, they can be controlled/negotiated
at the session set-up. Throughout this specification, these options
and variables are denoted as "parameters to be established through
out-of-band means". In Section 7, all of the parameters are
formally specified in the form of media type registration for the
AMR-WB+ encoding. 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 7.2 provides a
mapping of the parameters into the Session Description Protocol
(SDP) [6] for those applications that use SDP.
4. RTP Payload Format for AMR-WB+
The main emphasis in the payload design for AMR-WB+ has been to
minimize the overhead in typical use cases, while providing full
flexibility with a slightly higher overhead. In order to keep the
specification reasonably simple, we refrained from defining frame-
specific parameters for each frame type. Instead, a few common
parameters were specified that cover all types of frames.
The payload format has two modes, basic mode and interleaved mode.
The main structural difference between the two modes is the
extension of the table of content entries with frame displacement
fields (when operating in the interleaved mode). The basic mode
supports aggregation of multiple consecutive frames in a payload.
The interleaved mode supports aggregation of multiple frames that
are non-consecutive in time. In both modes it is possible to have
frames encoded with different frame types in the same payload. The
ISF must remain constant throughout the payload of a single packet.
The payload format is designed around the property of AMR-WB+ frames
that the frames are consecutive in time and share the same frame
duration (in the absence of an ISF change). This enables the
receiver to derive the timestamp for an individual frame within a
payload. In basic mode, the deriving process is based on the order
of frames. In interleaved mode, it is based on the compact
displacement fields. The frame timestamps are used to regenerate
the correct order of frames after reception, identify duplicates,
and detect lost frames that require concealment.
The interleaving scheme of this payload format is significantly more
flexible than the one specified in RFC 3267. The AMR and AMR-WB
payload format is only capable of using periodic patterns with
frames taken from an interleaving group at fixed intervals. The
interleaving scheme of this specification, in contrast, allows for
any interleaving pattern, as long as the distance in decoding order
between any two adjacent frames is not more than 256 frames. Note
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that even at the highest ISF this allows an interleaving depth up to
3.41 seconds.
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 audio frame multiple times. All
redundantly sent frames MUST use the same frame type and ISF, and
MUST have the same RTP timestamp, or MUST be a NO_DATA frame
(FT=15).
The payload consists of octet aligned elements (header, ToC and
audio frames). Only the audio frames for AMR-WB frame types (0-9)
require padding for octet alignment. If additional padding is
desired, then the P bit in the RTP header MAY be set and padding MAY
be appended as specified in [3].
4.1. RTP Header Usage
The format of the RTP header is specified in [3]. 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 in the packet. The timestamp
clock frequency SHALL be 72000 Hz. This frequency allows the frame
duration to be integer RTP timestamp ticks for the ISFs specified in
Table 1. It also provides reasonable conversion factors to the
input/output audio sampling frequencies supported by the codec. See
section 4.3.1 for guidance on how to derive the RTP timestamp for
any audio frame beyond the first one.
The RTP header marker bit (M) SHALL be set to 1 whenever the first
frame carried in the packet is the first frame in a talkspurt (see
definition of the talkspurt in section 4.1 of [9]). For all other
packets the marker bit SHALL be set to zero (M=0).
The assignment of an RTP payload type for the format defined in this
memo is outside the scope of this document. The RTP profile in use
either assigns a static payload type or mandates binding the payload
type dynamically.
The media type parameter "channels" is used to indicate the maximum
number of channels allowed for a given payload type. A payload type
where channels=1 (mono), SHALL only carry mono content. A payload
type for which channels=2 has been declared MAY carry both mono and
stereo content. Note that this definition is different from the one
in RFC 3551 [9]. As mentioned before, the AMR-WB+ codec handles the
support of stereo content and the (eventual) downmixing of stereo to
mono internally. This makes it unnecessary to negotiate for the
number of channels for reasons other than bit-rate efficiency.
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4.2. Payload Structure
The payload consists of a payload header, a table of contents, and
the audio data representing one or more audio frames. The following
diagram shows the general payload format layout:
+----------------+-------------------+----------------
| payload header | table of contents | audio data ...
+----------------+-------------------+----------------
Payloads containing more than one audio frame are called compound
payloads.
The following sections describe the variations taken by the payload
format depending on the mode in use, basic mode or interleaved mode.
4.3. Payload Definitions
4.3.1. Payload Header
The payload header carries data that is common for all frames in the
payload. The structure of the payload header is described below.
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
| ISF |TFI|L|
+-+-+-+-+-+-+-+-+
ISF (5 bits): Indicates the Internal Sampling Frequency employed for
all frames in this payload. The index value corresponds to
internal sampling frequency as specified in Table 24 in [1].
This field SHALL be set to 0 for payloads containing frames with
Frame Type values 0-13.
TFI (2 bits): Transport Frame Index, from 0 (first) to 3 (last),
indicating the position of the first transport frame of this
payload in the AMR-WB+ super-frame structure. For payloads with
frames of only Frame Type values 0-9 this field SHALL be set to
0. The TFI value for a frame of type 0-9 SHALL be ignored. Note
that the frame type is coded in the table of contents (as
discussed later) -- hence the mentioned dependencies of the frame
type can be applied easily by interpreting only values carried in
the payload header. It is not necessary to interpret the audio
bit stream itself.
L (1 bit): Long displacement field flag for payloads in interleaved
mode. If set to 0, four-bit displacement fields are used to
indicate interleaving offset; if set to 1, displacement fields of
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eight bits are used (see section 4.3.2.2). For payloads in the
basic mode this bit SHALL be set to 0 and SHALL be ignored by the
receiver.
Note that frames employing different ISF values require
encapsulation in separate packets. Thus, special considerations
apply when generating interleaved packets and an ISF change is
executed. In particular, frames that, according to the previously
used interleaving pattern, would be aggregated into a single packet
have to be separated into different packets, so that the
aforementioned condition (all frames in a packet share the ISF)
remains true. A naive implementation that splits the frames with
different ISF into different packets can result in up to twice the
number of RTP packets, when compared to an optimal interleaved
solution. Alteration of the interleaving before and after the ISF
change may reduce the need for extra RTP packets.
4.3.2. The Payload Table of Contents
The table of contents (ToC) consists of a list of entries, each
entry corresponds to a group of audio frames carried in the payload,
as depicted below.
+----------------+----------------+- ... -+----------------+
| ToC entry #1 | Toc entry #2 | ToC entry #N |
+----------------+----------------+- ... -+----------------+
When multiple groups of frames are present in a payload, the ToC
entries SHALL be placed in the packet in order of increasing RTP
timestamp value (modulo 2^32) of the first transport frame the TOC
entry represent.
4.3.2.1. ToC Entry in the Basic Mode
A ToC entry of a payload in the basic mode has the following format:
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F| Frame Type | #frames |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
F (1 bit): If set to 1, indicates that this ToC entry is followed by
another ToC entry; if set to 0, indicates that this ToC entry is
the last one in the ToC.
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Frame Type (FT) (7 bits): Indicates the audio codec frame type used
for the group of frames referenced by this ToC entry. FT
designates the combination of AMR-WB+ core and stereo rate, one
of the special AMR-WB+ frame types, the AMR-WB rate, or comfort
noise, as specified by Table 25 in [1].
#frames (8 bits): Indicates the number of frames in the group
referenced by this ToC entry. ToC entries with this field equal
to 0 (that would indicate zero frames) SHALL NOT be used and
received packets with such a TOC entry SHALL be discarded.
4.3.2.2. ToC Entry in the Interleaved Mode
Two different ToC entry formats are defined in interleaved mode.
They differ in the length of the displacement field, 4 bits or 8
bits. The L-bit in the payload header differentiates between the
two modes.
If L=0, a ToC entry has the following format:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F| Frame Type | #frames | DIS1 | ... | DISi | ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | ... | DISn | Padd |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
F (1 bit): See definition in 4.3.2.1.
Frame Type (FT) (7 bits): See definition in 4.3.2.1.
#frames (8 bits): See definition in 4.3.2.1.
DIS1...DISn (4 bits): A list of n (n=#frames) displacement fields
indicating the displacement of the i:th (i=1..n) audio frame
relative to the preceding audio frame in the payload, in units of
frames. The four-bit unsigned integer displacement values may be
between 0 and 15 indicating the number of audio frames in
decoding order between the (i-1):th and the i:th frame in the
payload. Note that for the first ToC entry of the payload the
value of DIS1 is meaningless. It SHALL be set to zero by a
sender, and SHALL be ignored by a receiver. This frame's location
in the decoding order is uniquely defined by the RTP timestamp
and TFI in the payload header. Note also that for subsequent ToC
entries DIS1 indicates the number of frames between the last
frame of the previous group and the first frame of this group.
Padd (4 bits): To ensure octet alignment, four padding bits SHALL be
included at the end of the ToC entry in case there is odd number
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of frames in the group referenced by this entry. These bits
SHALL be set to zero and SHALL be ignored by the receiver. If a
group containing an even number of frames is referenced by this
ToC entry, these padding bits SHALL NOT be included in the
payload.
If L=1, a ToC entry has the following format:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F| Frame Type | #frames | DIS1 | ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | DISn |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
F (1 bit): See definition in 4.3.2.1.
Frame Type (FT) (7 bits): See definition in 4.3.2.1.
#frames (8 bits): See definition in 4.3.2.1.
DIS1...DISn (8 bits): A list of n (n=#frames) displacement fields
indicating the displacement of the i:th (i=1..n) audio frame
relative to the preceding audio frame in the payload, in units of
frames. The eight-bit unsigned integer displacement values may
be between 0 and 255 indicating the number of audio frames in
decoding order between the (i-1):th and the i:th frame in the
payload. Note that for the first ToC entry of the payload the
value of DIS1 is meaningless. It SHALL be set to zero by a
sender, and SHALL be ignored by a receiver. This frame's location
in the decoding order is uniquely defined by the RTP timestamp
and TFI in the payload header. Note also that for subsequent ToC
entries DIS1 indicates the displacement between the last frame of
the previous group and the first frame of this group.
4.3.2.3. RTP Timestamp Derivation
The RTP Timestamp value for a frame SHALL be the timestamp value of
the first audio sample encoded in the frame. The timestamp value
for a frame is derived differently depending on the payload mode,
basic or interleaved. In both cases the first frame in a compound
packet has an RTP timestamp equal to the one received in the RTP
header. In the basic mode, the RTP time for any subsequent frame is
derived in two steps. First, the sum of the frame durations (see
Table 1) of all the preceding frames in the payload is calculated.
Then, this sum is added to the RTP header timestamp value. For
example, if the RTP Header timestamp value is 12345, the payload
carries four frames, and the frame duration is 16 ms (ISF = 32 kHz)
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corresponding to 1152 timestamp ticks, the RTP timestamp of the
fourth frame in the payload is 12345 + 3 * 1152 = 15801.
In interleaved mode, the RTP timestamp for each frame in the payload
is derived from the RTP header timestamp and the sum of the time
offsets of all preceding frames in this payload. The frame
timestamps are computed based on displacement fields and the frame
duration derived from the ISF value. Note that the displacement in
time between frame i-1 and frame i is (DISi + 1) * frame duration
because also the duration of the (i-1):th must be taken into
account. The timestamp of the first frame of the first group of
frames (TS(1)), i.e. the first frame of the payload is the RTP
header timestamp. For subsequent frames in the group the timestamp
is computed by
TS(i) = TS(i-1) + (DISi + 1) * frame duration, 2 < i < n
For subsequent groups of frames the timestamp of the first frame is
computed by
TS(1) = TSprev + (DIS1 + 1) * frame duration,
where TSprev denotes the timestamp of the last frame in the previous
group. The timestamps of the subsequent frames in the group are
computed in the same way as for the first group.
The following example derives the RTP timestamps for the frames in
an interleaved mode payload having the following header and ToC
information:
RTP header timestamp: 12345
ISF = 32 kHz
Frame 1 displacement field: DIS1 = 0
Frame 2 displacement field: DIS2 = 6
Frame 3 displacement field: DIS3 = 4
Frame 4 displacement field: DIS4 = 7
Assuming an ISF of 32 kHz, which implies frame duration of 16 ms,
one frame lasts 1152 ticks. The timestamp of the first frame in the
payload is the RTP timestamp, i.e. TS(1) = RTP TS. Note that the
displacement field value for this frame must be ignored. For the
second frame in the payload the timestamp can be calculated as TS(2)
= TS(1) + (DIS2 + 1) * 1152 = 20409. For the third frame the
timestamp is TS(3) = TS(2) + (DIS3 + 1) * 1152 = 26169. Finally,
for the fourth frame of the payload we have TS(4) = TS(3) + (DIS4 +
1) * 1152 = 35385.
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4.3.2.4. Frame Type Considerations
The value of Frame Type (FT) is defined in Table 25 in [1]. FT=14
(AUDIO_LOST) is used to denote frames that are lost. A NO_DATA
(FT=15) frame could be the result of two conditions: First, to
indicate that no data has been produced by the audio encoder, and
second that no data is transmitted in the current payload. An
example for the latter would be that the frame in question has been
or will be sent in an earlier or later packet. The duration for
these non-included frames is dependent on the internal sampling
frequency indicated by the ISF field.
For frame types with index 0-13 the ISF field SHALL be set 0. The
frame duration for these frame types is fixed to 20 ms in time, i.e.
1440 ticks in 72 kHz. For payloads containing only frames of type
0-9, the TFI field SHALL be set to 0, and SHALL be ignored by the
receiver. In a payload combining frames of type 0-9 and 10-13 the
TFI values needs to be set to match the transport frames of type 10-
13. Thus, frames of type 0-9 will also have a derived TFI, which is
ignored.
4.3.2.5. Other TOC Considerations
If a ToC entry with an undefined FT value is received, the whole
packet SHALL be discarded. This is to avoid the loss of data
synchronization in the depacketization process, which can result in
a severe degradation in audio quality.
Packets containing only NO_DATA frames SHOULD NOT be transmitted.
Also, NO_DATA frames at the end of a frame sequence to be carried in
a payload SHOULD NOT be included in the transmitted packet. The
AMR-WB+ SCR/DTX is identical with AMR-WB SCR/DTX described in [5]
and can only be used in combination with the AMR-WB frame types (0-
8).
When multiple groups of frames are present, their ToC entries SHALL
be placed in the ToC in the order of increasing RTP timestamp value
(modulo 2^32) of the first transport frame the TOC entry represents,
independent of the payload mode. In basic mode the frames SHALL be
consecutive in time, while in interleaved mode the frames MAY not
only be non-consecutive in time but MAY even have varying inter
frame distances.
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4.3.2.6. ToC Examples
The following example illustrates a ToC for three audio frames in
basic mode. Note that in this case all audio frames are encoded
using the same frame type, i.e. there is only one ToC entry.
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| Frame Type1 | #frames = 3 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The next example depicts a ToC of three entries in basic mode. Note
that in this case the payload carries also three frames, but three
ToC entries are needed because the frames of the payload are encoded
using different frame types.
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| Frame Type1 | #frames = 1 |1| Frame Type2 | #frames = 1 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| Frame Type3 | #frames = 1 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The following example illustrates a ToC with two entries in
interleaved mode using four bit displacement fields. The payload
includes two groups of frames, the first one including a single
frame, and the other one consisting of two frames.
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| Frame Type1 | #frames = 1 | DIS1 | padd |0| Frame Type2 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| #frames = 2 | DIS1 | DIS2 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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4.3.3. Audio Data
Audio data of a payload consists of zero or more audio frames, as
described in the ToC of the payload.
ToC entries with FT=14 or 15 represent frame types with a length of
0. Hence, no data SHALL be placed in the audio data section to
represent frames of this type.
As already discussed before, each audio frame of an extension frame
type represents an AMR-WB+ transport frame corresponding to the
encoding of 512 samples of audio, sampled with the internal sampling
frequency specified by the ISF indicator. As an exception, frame
types with index 10-13 are only capable of using a single internal
sampling frequency (25600 Hz). The encoding rates (combination of
core bit-rate and stereo bit-rate) are indicated in the frame type
field of the corresponding ToC entry. The octet length of the audio
frame is implicitly defined by the frame type field and is given in
tables 21 and 25 of [1]. The order and numbering notation of the
bits are as specified in [1]. For the AMR-WB+ extension frame types
and comfort noise frames, the bits are in the order produced by the
encoder. The last octet of each audio frame MUST be padded with
zeroes at the end if not all bits in the octet are used. In other
words, each audio frame MUST be octet-aligned.
4.3.4. Methods for Forming the Payload
The payload begins with the payload header, followed by the table of
contents that consists of a list of ToC entries.
The audio data follows the table of contents. All of the octets
comprising an audio frame SHALL be appended to the payload as a
unit. The audio frames are packetized in timestamp order within
each group of frames (per ToC entry). The groups of frames are
packetized in the same order as their corresponding ToC entries.
Note that there are no data octets in a group having a ToC entry
with FT=14 or FT=15.
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4.3.5. Payload Examples
4.3.5.1. Example 1, Basic Mode Payload Carrying Multiple Frames Encoded
Using the Same Frame Type
Figure 4 depicts a payload that carries three AMR-WB+ frames encoded
using 14 kbit/s frame type (FT=26) with a frame length of 280 bits
(35 bytes). The internal sampling frequency in this example is 25.6
kHz (ISF = 8). The TFI for the first frame is 2, indicating that
the first transport frame in this payload is the third in a super-
frame. Since this payload is in the basic mode the subsequent
frames of the payload are consecutive frames in decoding order, i.e.
the fourth transport frame of the current super-frame and the first
transport frame of the next super-frame. Note that because the
frames are all encoded using the same frame type, only one ToC entry
is required.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ISF = 8 | 2 |0|0| FT = 26 | #frames = 3 | f1(0...7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f1(272...279) | f2(0...7) | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| f2(272...279) | f3(0...7) | ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f3(272...279) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 4: An example of a basic mode payload carrying three frames
of the same frame type.
4.3.5.2. Example 2, Basic Mode Payload Carrying Multiple Frames Encoded
Using Different Frame Types
Figure 5 depicts a payload that carries three AMR-WB+ frames; the
first frame is encoded using 18.4 kbit/s frame type (FT=33) with a
frame length of 368 bits (46 bytes), and the two subsequent frames
are encoded using 20 kbit/s frame type (FT=35) having frame length
of 400 bits (50 bytes). The internal sampling frequency in this
example is 32 kHz (ISF = 10), implying the overall bit-rates of 23
kbit/s for the first frame of the payload, and 25 kbit/s for the
subsequent frames. The TFI for the first frame is 3, indicating
that the first transport frame in this payload is the fourth in a
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super-frame. Since this is a payload in the basic mode, the
subsequent frames of the payload are consecutive frames in decoding
order, i.e. the first and second transport frames of the current
super-frame. Note that since the payload carries two different
frame types, there are two ToC entries.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ISF=10 | 3 |0|1| FT = 33 | #frames = 1 |0| FT = 35 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| #frames = 2 | f1(0...7) | ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f1(360...367) | f2(0...7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| f2(392...399) | f3(0...7) | ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f3(392...399) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 5: An example of a basic mode payload carrying three frames
employing two different frame types.
4.3.5.3. Example 3, Payload in Interleaved Mode
The example in Figure 6 depicts a payload in interleaved mode,
carrying four frames encoded using 32 kbit/s frame type (FT=47) with
frame length of 640 bits (80 bytes). The internal sampling
frequency is 38.4 kHz (ISF = 13), implying a bit-rate of 48 kbit/s
for all frames in the payload. The TFI for the first frame is 0,
hence it is the first transport frame of a super-frame. The
displacement fields for the subsequent frames are DIS2=18, DIS3=15,
and DIS4=10, which indicates that the subsequent frames have the
TFIs of 3, 3, and 2, respectively. The long displacement field flag
L in the payload header is set to 1, which results in the use of
eight bits for the displacement fields in the ToC entry. Note that
since all frames of this payload are encoded using the same frame
type, there is need only for a single ToC entry. Furthermore, the
displacement field for the first frame (corresponding to the first
ToC entry with DIS1=0) must be ignored, since its timestamp and TFI
are defined by the RTP timestamp and the TFI found in the payload
header.
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The RTP timestamp values of the frames in this example is:
Frame1: TS1 = RTP Timestamp
Frame2: TS2 = TS1 + 19 * 960
Frame3: TS3 = TS2 + 16 * 960
Frame4: TS4 = TS3 + 11 * 960
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ISF=13 | 0 |1|0| FT = 47 | #frames = 4 | DIS1 = 0 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| DIS2 = 18 | DIS3 = 15 | DIS4 = 10 | f1(0...7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f1(632...639) | f2(0...7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f2(632...639) | f3(0...7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f3(632...639) | f4(0...7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f4(632...639) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 6: An example of an interleaved mode payload carrying four
frames at the same frame type.
4.4. Interleaving Considerations
The use of interleaving requires further considerations. As
presented in the example in Section 3.6.2, a given interleaving
pattern requires a certain amount of the deinterleaving buffer.
This buffer space, expressed in a number of transport frame slots,
is indicated by the "interleaving" media parameter. The number of
frame slots needed can be converted into actual memory requirements
by considering the 80 bytes per frame used by the largest
combination of AMR-WB+'s core and stereo rates.
The information about the frame buffer size is not always sufficient
to determine when it is appropriate to start consuming frames from
the interleaving buffer. There are two cases in which additional
information is needed: first, when switching of the ISF occurs, and
second when the interleaving pattern changes. The "int-delay" media
type parameter is defined to convey this information. It allows a
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sender to indicate the minimal media time that needs to be present
in the buffer before the decoder can start consuming frames from the
buffer. Because the sender has full control over ISF changes and
the interleaving pattern, it can calculate this value.
In certain cases, for example if joining a multicast session with
interleaving mid-session, a receiver may initially receive only part
of the packets in the interleaving pattern. This initial partial
reception (in frame sequence order) of frames can yield too few
frames for acceptable quality from the audio decoding. This problem
also arises when using encryption for access control, and the
receiver does not have the previous key.
Although the AMR-WB+ is robust and thus tolerant to a high random
frame erasure rate, it would have difficulties handling consecutive
frame losses at startup. Thus some special implementation
considerations are described. In order to efficiently handle this
type of startup, it must be noted that decoding is only possible to
start at the beginning of a super-frame, and that holds true even if
the first transport frame is indicated as lost. Secondly, decoding
is only RECOMMENDED to start if at least 2 transport frames are
available out of the 4 belonging to that super-frame.
After receiving a number of packets, in the worst case as many
packets as the interleaving pattern covers, the previously described
effects disappear and normal decoding is resumed.
Similar issues arise when a receiver leaves a session or has lost
access to the stream. In the case of the receiver leaving the
session, this would be a minor issue since playout is normally
stopped. It is also a minor issue for the case of lost access, since
the AMR-WB+ error concealment will fade out the audio if massive
consecutive losses are encountered.
The sender can avoid this type of problems in many sessions by
starting and ending interleaving patterns correctly when risks of
losses occur. One such example is a key-change done for access
control to encrypted streams. If only some keys are provided to
clients and there is a risk of them receiving content for which they
do not have the key, it is recommended that interleaving patterns
not overlap key changes.
4.5. Implementation Considerations
An application implementing this payload format MUST understand all
the payload parameters. Any mapping of the parameters to a
signaling protocol MUST support all parameters. So an
implementation of this payload format in an application using SDP is
required to understand all the payload parameters in there SDP-
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mapped form. This requirement ensures that an implementation always
can decide whether it is capable to communicate.
Both basic and interleaving mode SHALL be implemented. The
implementation burden of both is rather small and requiring both
ensures interoperability. As the AMR-WB+ codec contains the full
functionality of the AMR-WB codec, it is RECOMMENDED to also
implement the payload format in RFC 3267 [7] for the AMR-WB frame
types when implementing this specification. Doing so makes the
interoperability with devices that only support AMR-WB more likely.
The switching of ISF combined with packet loss could result in
concealment using the wrong audio frame length. This can occur if
packet loss(es) result in lost frames directly after the point of
ISF change. The packet loss would prevent the receiver from
noticing the changed ISF and thereby conceal the lost transport
frame with the previous ISF, instead of the new one. Such an error,
although always later detectable results in boundary misalignment,
which can cause audio distortions and problems with synchronization,
as too many or too few audio samples were created. This problem can
be mitigated in most cases by performing ISF recovery prior to
concealment as outlined in section 4.5.1 below.
4.5.1. ISF recovery in case of packet loss
In case of packet loss, it is important that the AMR-WB+ decoder
initiates a proper error concealment to replace the frames carried
in the lost packet. A loss concealment algorithm requires a codec
framing that matches the timestamps of the correctly received
frames. Hence, it is necessary to recover the timestamps of the
lost frames. Doing in so is non-trivial because the codec frame
length that is associated with the ISF may have changed during the
frame loss.
In the following, the recovery of the timestamp information of lost
frames is illustrated by the means of an example. Two frames with
timestamps t0 and t1 have been received properly, the first one
being the last packet before the loss, and the latter one is the
first packet after the loss period. The ISF values for these
packets are isf0 and isf1, respectively. The TFIs of these frames
are tfi0 and tfi1, respectively. The associated frame lengths (in
timestamp ticks) are given as L0 and L1, respectively. In this
example three frames with timestamps x1 - x3 have been lost. The
example further assumes that ISF changes once from isf0 to isf1
during the frame loss period, as shown in the figure below.
Since not all information required for the full recovery of the
timestamps is generally known in the receiver, an algorithm is
needed to estimate the ISF associated with the lost frames. Also
the number of lost frames needs to be recovered.
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|<---L0--->|<---L0--->|<-L1->|<-L1->|<-L1->|
| Rxd | lost | lost | lost | Rxd |
--+----------+----------+------+------+------+--
t0 x1 x2 x3 t1
Example Algorithm:
Start: # check for frame loss
If (t0 + L0) == t1 Then goto End # no frame loss
Step 1: # check case with no ISF change
If (isf0 != isf1) Then goto Step 2 # At least one ISF change
If (isFractional(t1 - t0)/L0) Then goto Step 3
# More than 1 ISF change
Return recovered timestamps as
x(n) = t0 + n*L1 and associated ISF equal to isf0, for 0<n<(t1 -
t0)/L0
goto End
Step 2:
Loop initialization: n := 4 - tfi0 mod 4
While n <= (t1-t0)/L0
Evaluate m := (t1 - t0 - n*L0)/L1
If (isInteger(m) AND ((tfi0+n+m) mod 4 == tfi1)) Then goto found;
n := n+4
endloop
goto step 3 # More than 1 ISF change
found:
Return recovered timestamps and ISFs as
x(i) = t0 + i*L0 and associated ISF equal to isf0, for 0 < i <= n
x(i) = t0 + n*L0 + (i-n)*L1 and associated ISF equal to isf1, for n
< i <= n+m
goto End
Step 3:
More than 1 ISF change has occurred. Since ISF changes can be
assumed to be infrequent, such a situation occurs only if long
sequences of frames are lost. In that case it is probably not
useful to try to recover the timestamps of the lost frames. Rather,
the AMR-WB+ decoder should be reset and decoding should be resumed
starting with the frame with timestamp t1.
End:
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The above algorithm does still not solve the issue when the receiver
buffer depth is shallower than the loss burst. In this kind of case
where the concealment must be done without any knowledge about
future frames, the concealment may result in loss of frame boundary
alignment. If that occurs, it may be necessary to reset and restart
the codec to perform resynchronization.
4.5.2. Decoding Validation
If the receiver finds a mismatch between the size of a received
payload and the size indicated by the ToC of the payload, the
receiver SHOULD discard the packet. This is recommended because
decoding a frame parsed from a payload based on erroneous ToC data
could severely degrade the audio quality.
5. Congestion Control
The general congestion control considerations for transporting RTP
data apply, see RTP [3] and any applicable RTP profile like AVP [9].
However, the multi-rate capability of AMR-WB+ audio coding provides
a mechanism that may help to control congestion, since the bandwidth
demand can be adjusted (within the limits of the codec) by selecting
a different coding frame type or lower internal sampling rate.
The number of frames encapsulated in each RTP payload highly
influences the overall bandwidth of the RTP stream due to header
overhead constraints. Packetizing more frames in each RTP payload
can reduce the number of packets sent and hence the header overhead,
at the expense of increased delay and reduced error robustness.
If forward error correction (FEC) is used, the amount of FEC-induced
redundancy needs to be regulated such that the use of FEC itself
does not cause a congestion problem.
6. Security Considerations
RTP packets using the payload format defined in this specification
are subject to the general security considerations discussed in RTP
[3] and any applicable profile such as AVP [9] or SAVP [10]. As
this format transports encoded audio, the main security issues
include confidentiality, integrity protection, and data origin
authentication of the audio itself. The payload format itself does
not have any built-in security mechanisms. Any suitable external
mechanisms, such as SRTP [10], MAY be used.
This payload format, or the AMR-WB+ decoder, do not exhibit any
significant non-uniformity in the receiver side computational
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complexity for packet processing, and thus are unlikely to pose a
denial-of-service threat due to the receipt of pathological data.
6.1. Confidentiality
In order to ensure confidentiality of the encoded audio, all audio
data bits MUST be encrypted. There is less need to encrypt the
payload header or the table of contents since they only carry
information about the frame type. This information could also be
useful to a third party, for example for quality monitoring.
The use of interleaving in conjunction with encryption can have a
negative impact on the confidentiality, for a short period of time.
Consider the following packets (in brackets) containing frame
numbers as indicated: {10, 14, 18}, {13, 17, 21}, {16, 20, 24} (a
popular continuous diagonal interleaving pattern). The originator
wishes to deny some participants the ability to hear material
starting at time 16. Simply changing the key on the packet with the
timestamp at or after 16, and denying that new key to those
participants, does not achieve this; frames 17, 18 and 21 have been
supplied in prior packets under the prior key, and error concealment
may make the audio intelligible at least as far as frame 18 or 19,
and possibly further.
6.2. Authentication and Integrity
To authenticate the sender of the speech, an external mechanism MUST
be used. It is RECOMMENDED that such a mechanism protects both the
complete RTP header and the payload (speech and data bits).
Data tampering by a man-in-the-middle attacker could replace audio
content and also result in erroneous depacketization/decoding that
could lower the audio quality.
7. Payload Format Parameters
This section defines the parameters that may be used to select
features of the AMR-WB+ payload format. The parameters are defined
as part of the media type registration for the AMR-WB+ audio codec.
A mapping of the parameters into the Session Description Protocol
(SDP) [6] is also provided for those applications that use SDP.
Equivalent parameters could be defined elsewhere for use with
control protocols that do not use MIME or SDP.
The data format and parameters are only specified for real-time
transport in RTP.
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7.1. Media Type Registration
The media type for the Extended Adaptive Multi-Rate Wideband (AMR-
WB+) codec is allocated from the IETF tree since AMR-WB+ is expected
to be a widely used audio codec in general streaming applications.
Note: parameters not listed below MUST be ignored by the receiver.
Media Type name: audio
Media subtype name: AMR-WB+
Required parameters:
None
Optional parameters:
channels: The maximum number of audio channels used by the
audio frames. Permissible values are 1 (mono) or 2
(stereo). If no parameter is present, the maximum
number of channels is 2 (stereo). Note: when set to
1, implicitly the stereo frame types cannot be used.
interleaving: Indicates that frame level interleaving mode SHALL
be used for the payload. The parameter specifies
the number of transport frame slots required in a
deinterleaving buffer (including the frame that is
ready to be consumed). Its value is equal to one
plus the maximum number of frames that precede any
frame in transmission order and follow the frame in
RTP timestamp order. The value MUST be greater than
zero. If this parameter is not present,
interleaving mode SHALL NOT be used.
int-delay: The minimal media time delay in RTP timestamp ticks
that is needed in the deinterleaving buffer, i.e.
the difference in RTP timestamp ticks between the
earliest and latest audio frame present in the
deinterleaving buffer.
ptime: see section 6 in RFC2327 [6].
maxptime: see Section 8 in RFC 3267 [7].
Restriction on Usage:
This type is only defined for transfer via RTP (STD
64).
Encoding considerations:
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An RTP payload according to this format is binary data
and thus may need to be appropriately encoded in non-
binary environments. However as long as used within
RTP, no encoding is necessary.
Security considerations:
See Section 6 of RFC XXXX.
Interoperability considerations:
To maintain interoperability with AMR-WB capable end-
points, in cases where negotiation is possible and the
AMR-WB+ end-point supporting this format also supports
RFC 3267 for AMR-WB transport, an AMR-WB+ end-point
SHOULD declare itself also as AMR-WB capable (i.e.
supporting also "audio/AMR-WB" as specified in RFC
3267).
As the AMR-WB+ decoder is capable of performing stereo
to mono conversions, all receivers of AMR-WB+ should be
able to receive both stereo and mono, although the
receiver only is capable of playout of mono signals.
Public specification:
RFC XXXX
3GPP TS 26.290, see reference [1] of RFC XXXX
Additional information:
This MIME type is not applicable for file storage.
Instead file storage of AMR-WB+ encoded audio is
specified within the 3GPP defined ISO based multimedia
file format defined in 3GPP TS 26.244, see reference
[14] of RFC XXXX. This file format has the MIME types
"audio/3GPP" or "video/3GPP" as defined by RFC 3839
[15].
Person & email address to contact for further information:
magnus.westerlund@ericsson.com
ari.lakaniemi@nokia.com
Intended usage: COMMON.
It is expected that many IP based streaming
applications will use this type.
Change controller:
IETF Audio/Video Transport working group delegated from
the IESG.
7.2. Mapping Media Type Parameters into SDP
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The information carried in the media type specification has a
specific mapping to fields in the Session Description Protocol (SDP)
[6], which is commonly used to describe RTP sessions. When SDP is
used to specify RTP session using this RTP payload format, the
mapping is as follows:
- The media type ("audio") is used in SDP "m=" as the media name.
- The media type (payload format name) is used in SDP "a=rtpmap" as
the encoding name. The RTP clock rate in "a=rtpmap" SHALL be
72000 for AMR-WB+, and the encoding parameter number of channels
MUST either be explicitly set to 1 or 2, or be omitted, implying
the default value of 2.
- The parameters "ptime" and "maxptime" are placed in the SDP
attributes "a=ptime" and "a=maxptime", respectively.
- Any remaining parameters are placed in the SDP "a=fmtp" attribute
by copying them directly from the MIME media type string as a
semicolon separated list of parameter=value pairs.
7.2.1. Offer-Answer Model Considerations
To achieve good interoperability in an Offer-Answer [8] negotiation
usage, the following considerations should be taken into account:
For negotiable offer/answer usage the following interpretation rules
SHALL be applied:
- The "interleaving" parameter is symmetric, thus requiring that
the answerer must also include it for the answer to an offered
payload type which contains the parameter. However, the buffer
space value is declarative in usage in unicast. For multicast
usage the same value in the response is required in order to
accept the payload type. For streams declared as sendrecv or
recvonly: The receiver will accept reception of streams using the
interleaved mode of the payload format. The value declares the
amount of buffer space the receiver has available for the sender
to utilize. For sendonly streams the parameter indicates the
desired configuration and amount of buffer space. An answerer is
RECOMMENDED to respond using the offered value, if capable of
using it.
- The "int-delay" parameter is declarative. For streams declared
as sendrecv or recvonly the value indicate the maximum initial
delay the receiver will accept in the deinterleaving buffer. For
sendonly streams the value is the amount of media time the sender
desires to use, the value SHOULD be copied into any response.
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- The "channels" parameter is declarative. For "sendonly" streams
it indicates the desired channel usage, stereo and mono, or mono
only. For "recvonly" and "sendrecv" streams the parameter
indicates what the receiver accepts to use. As any receiver will
be capable of receiving stereo frame type and perform local
mixing within the AMR-WB+ decoder, there is normally only one
reason to restrict to mono only: to avoid spending bit-rate on
data that are not utilized if the front-end is only capable of
mono.
- The "ptime" parameter works as indicated by the offer/answer
model [8], "maxptime" SHALL be used in the same way.
- To maintain interoperability with AMR-WB in cases where
negotiation is possible, an AMR-WB+ capable end-point which also
implements the AMR-WB payload format [7] is RECOMMENDED to also
declare itself capable of AMR-WB as it is a subset of the AMR-WB+
codec.
In declarative usage, like SDP in RTSP [16] or SAP [17], the
following interpretation of the parameters SHALL be done:
- The "interleaving" parameter, if present, configures the payload
format in that mode, and the value indicates the number of frames
that the deinterleaving buffer is required to support to be able
to handle this session correctly.
- The "int-delay" parameter indicates the initial buffering delay
required to receive this stream correctly.
- The "channels" parameter indicates if the content being
transmitted can contain either both stereo and mono rates, or
only mono.
- All other parameters indicate values that are being used by the
sending entity.
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7.2.2. Examples
One example SDP session description utilizing AMR-WB+ mono and
stereo encoding follow.
m=audio 49120 RTP/AVP 99
a=rtpmap:99 AMR-WB+/72000/2
a=fmtp:99 interleaving=30; int-delay=86400
a=maxptime:100
Note that the payload format (encoding) names are commonly shown in
upper case. Media 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.
8. IANA Considerations
It is requested that one new MIME subtype (audio/amr-wb+) is
registered by IANA, see Section 7.
9. Contributors
Daniel Enstrom has contributed in writing the codec introduction
section. Stefan Bruhn has contributed by writing the ISF recovery
algorithm.
10. Acknowledgements
The authors would like to thank Redwan Salami and Stefan Bruhn for
their significant contributions made throughout the writing and
reviewing of this document. Dave Singer contributed by reviewing
and suggesting improved language. Anisse Taleb and Ingemar
Johansson contributed by implementing the payload format, and thus
helped locating some flaws. We would also like to acknowledge
Qiaobing Xie, coauthor of RFC 3267 on which this document is based
on.
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11. References
11.1. Normative references
[1] 3GPP TS 26.290 "Audio codec processing functions; Extended
Adaptive Multi-Rate Wideband (AMR-WB+) codec; Transcoding
functions", version 6.1.0 (2004-12), 3rd Generation Partnership
Project (3GPP).
[2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, Internet Engineering Task Force,
March 1997.
[3] H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, "RTP: A
Transport Protocol for Real-Time Applications", STD 64, RFC
3550, Internet Engineering Task Force, July 2003.
[4] 3GPP TS 26.192 "AMR Wideband speech codec; Comfort Noise
aspects", version 5.0.0 (2001-03), 3rd Generation Partnership
Project (3GPP).
[5] 3GPP TS 26.193 "AMR Wideband speech codec; Source Controlled
Rate operation", version 5.0.0 (2001-03), 3rd Generation
Partnership Project (3GPP).
[6] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, Internet Engineering Task Force, April
1998.
[7] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie, "Real-
Time Transport Protocol (RTP) Payload Format and File Storage
Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-
Rate Wideband (AMR-WB) Audio Codecs", RFC 3267, Internet
Engineering Task Force, June 2002.
[8] J. Rosenberg, and H. Schulzrinne, "An Offer/Answer Model with
the Session Description Protocol (SDP)", RFC 3264, Internet
Engineering Task Force, June 2002.
[9] Schulzrinne, H., Casner S., "RTP Profile for Audio and Video
Conferences with Minimal Control", STD 65, RFC 3551, Internet
Engineering Task Force, July 2003.
Sjoberg, et al. Standards Track [Page 35]
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11.2. Informative references
[10] Baugher, et. al., "The Secure Real Time Transport Protocol",
RFC 3711, Internet Engineering Task Force, March 2004.
[11] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for
Generic Forward Error Correction", RFC 2733, Internet
Engineering Task Force, December 1999.
[12] 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, Internet
Engineering Task Force, September 1997.
[13] 3GPP TS 26.233 "Packet Switched Streaming service", version
5.0.0 (2001-03), 3rd Generation Partnership Project (3GPP).
[14] 3GPP TS 26.244 " Transparent end-to-end packet switched
streaming service (PSS); 3GPP file format (3GP)", version 6.1.0
(2004-09), 3rd Generation Partnership Project (3GPP).
[15] D. Singer, and R. Castagno, "MIME Type Registrations for 3rd
Generation Partnership Project (3GPP) Multimedia files," RFC
3839, Internet Engineering Task Force, July 2004.
[16] H. Schulzrinne, A. Rao, R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, Internet Engineering Task Force,
April 1998.
[17] M. Handley, C. Perkins, E. Whelan, "Session Announcement
Protocol", RFC 2974, Internet Engineering Task Force, June
2001.
Any 3GPP document can be downloaded from the 3GPP webserver,
"http://www.3gpp.org/", see specifications.
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12. Authors' Addresses
Johan Sjoberg
Ericsson Research
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
Stephan Wenger
Nokia Corporation
P.O. Box 100
FIN-33721 Tampere
FINLAND
Phone: +358-50-486-0637
EMail: Stephan.Wenger@nokia.com
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13. IPR Notice
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Intellectual Property Rights or other rights that might be claimed
to pertain to the implementation or use of the technology described
in this document or the extent to which any license under such
rights might or might not be available; nor does it represent that
it has made any independent effort to identify any such rights.
Information on the procedures with respect to rights in RFC
documents can be found in BCP 78 and BCP 79.
Copies of IPR disclosures made to the IETF Secretariat and any
assurances of licenses to be made available, or the result of an
attempt made to obtain a general license or permission for the use
of such proprietary rights by implementers or users of this
specification can be obtained from the IETF on-line IPR repository
at http://www.ietf.org/ipr.
The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights that may cover technology that may be required to implement
this standard. Please address the information to the IETF at ietf-
ipr@ietf.org.
14. Copyright Notice
Copyright (C) The Internet Society (2005). This document is subject
to the rights, licenses and restrictions contained in BCP 78, and
except as set forth therein, the authors retain all their rights.
This document and the information contained herein are provided on
an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE
REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE
INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
This Internet-Draft expires in March 2006.
RFC Editor Considerations
The RFC editor is requested to replace all occurrences of XXXX with
the RFC number this document receives.
Sjoberg, et al. Standards Track [Page 38]
