| RFC : | rfc6366 |
| Title: | |
| Date: | August 2011 |
| Status: | INFORMATIONAL |
Internet Engineering Task Force (IETF) J. Valin
Request for Comments: 6366 Mozilla
Category: Informational K. Vos
ISSN: 2070-1721 Skype Technologies, S.A.
August 2011
Requirements for an Internet Audio Codec
Abstract
This document provides specific requirements for an Internet audio
codec. These requirements address quality, sampling rate, bit-rate,
and packet-loss robustness, as well as other desirable properties.
Status of This Memo
This document is not an Internet Standards Track specification; it is
published for informational purposes.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Not all documents
approved by the IESG are a candidate for any level of Internet
Standard; see Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc6366.
Copyright Notice
Copyright (c) 2011 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
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Table of Contents
1. Introduction ....................................................2
2. Definitions .....................................................3
3. Applications ....................................................3
3.1. Point-to-Point Calls .......................................3
3.2. Conferencing ...............................................4
3.3. Telepresence ...............................................5
3.4. Teleoperation and Remote Software Services .................5
3.5. In-Game Voice Chat .........................................5
3.6. Live Distributed Music Performances / Internet
Music Lessons ..............................................6
3.7. Delay-Tolerant Networking or Push-to-Talk Services .........6
3.8. Other Applications .........................................7
4. Constraints Imposed by the Internet on the Codec ................7
5. Detailed Basic Requirements .....................................8
5.1. Operating Space ............................................9
5.2. Quality and Bit-Rate .......................................9
5.3. Packet-Loss Robustness ....................................10
5.4. Computational Resources ...................................10
6. Additional Considerations ......................................12
6.1. Low-Complexity Audio Mixing ...............................12
6.2. Encoder Side Potential for Improvement ....................12
6.3. Layered Bit-Stream ........................................13
6.4. Partial Redundancy ........................................13
6.5. Stereo Support ............................................13
6.6. Bit Error Robustness ......................................13
6.7. Time Stretching and Shortening ............................14
6.8. Input Robustness ..........................................14
6.9. Support of Audio Forensics ................................14
6.10. Legacy Compatibility .....................................14
7. Security Considerations ........................................14
8. Acknowledgments ................................................15
9. Informative References .........................................15
1. Introduction
This document provides requirements for an audio codec designed
specifically for use over the Internet. The requirements attempt to
address the needs of the most common Internet interactive audio
transmission applications and ensure good quality when operating in
conditions that are typical for the Internet. These requirements
also address the quality, sampling rate, delay, bit-rate, and packet-
loss robustness. Other desirable codec properties are considered as
well.
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2. Definitions
Throughout this document, the following conventions refer to the
sampling rate of a signal:
Narrowband: 8 kilohertz (kHz)
Wideband: 16 kHz
Super-wideband: 24/32 kHz
Full-band: 44.1/48 kHz
Codec bit-rates in bits per second (bit/s) will be considered without
counting any overhead ((IP/UDP/RTP) headers, padding, etc.). The
codec delay is the total algorithmic delay when one adds the codec
frame size to the "look-ahead". Thus, it is the minimum
theoretically achievable end-to-end delay of a transmission system
that uses the codec.
3. Applications
The following applications should be considered for Internet audio
codecs, along with their requirements:
o Point-to-point calls
o Conferencing
o Telepresence
o Teleoperation
o In-game voice chat
o Live distributed music performances / Internet music lessons
o Delay-tolerant networking or push-to-talk services
o Other applications
3.1. Point-to-Point Calls
Point-to-point calls are voice over IP (VoIP) calls from two
"standard" (fixed or mobile) phones, and implemented in hardware or
software. For these applications, a wideband codec is required,
along with narrowband support for compatibility with a public
switched telephone network (PSTN). It is expected for the range of
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useful bit-rates to be 12 - 32 kilobits per second (kbit/s) for
wideband speech and 8 - 16 kbit/s for narrowband speech. The codec
delay must be less than 40 milliseconds (ms), but no more than 25 ms
is desirable. Support for encoding music is not required, but it is
desirable for the codec not to make background (on-hold) music
excessively unpleasant to hear. Also, the codec should be robust to
noise (produce intelligible speech and no annoying artifacts) even at
lower bit-rates.
3.2. Conferencing
Conferencing applications (that support multi-party calls) have
additional requirements on top of the requirements for point-to-point
calls. Conferencing systems often have higher-fidelity audio
equipment and have greater network bandwidth available -- especially
when video transmission is involved. Therefore, support for super-
wideband audio becomes important, with useful bit-rates in the 32 -
64 kbit/s range. The ability to vary the bit-rate, according to the
"difficulty" of the audio signal, is a desirable feature for the
codec. This not only saves bandwidth "on average", but it can also
help conference servers make more efficient use of the available
bandwidth, by using more bandwidth for important audio streams and
less bandwidth for less important ones (e.g., background noise).
Conferencing end-points often operate in hands-free conditions, which
creates acoustic echo problems. Therefore, lower delay is important,
as it reduces the quality degradation due to any residual echo after
acoustic echo cancellation (AEC). Consequently, the codec delay must
be less than 30 ms for this application. An optional low-delay mode
with less than 10 ms delay is desirable, but not required.
Most conferencing systems operate with a bridge that mixes some (or
all) of the audio streams and sends them back to all the
participants. In that case, it is important that the codec not
produce annoying artifacts when two voices are present at the same
time. Also, this mixing operation should be as easy as possible to
perform. To make it easier to determine which streams have to be
mixed (and which are noise/silence), it must be possible to measure
(or estimate) the voice activity in a packet without having to fully
decode the packet (saving most of the complexity when the packet need
not be decoded). Also, the ability to save on the computational
complexity when mixing is also desirable, but not required. For
example, a transform codec may make it possible to mix the streams in
the transform domain, without having to go back to time-domain. Low-
complexity up-sampling and down-sampling within the codec is also a
desirable feature when mixing streams with different sampling rates.
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3.3. Telepresence
Most telepresence applications can be considered to be essentially
very high-quality video-conferencing environments, so all of the
conferencing requirements also apply to telepresence. In addition,
telepresence applications require super-wideband and full-band audio
capability with useful bit-rates in the 32 - 80 kbit/s range. While
voice is still the most important signal to be encoded, it must be
possible to obtain good quality (even if not transparent) music.
Most telepresence applications require more than one audio channel,
so support for stereo and multi-channel is important. While this can
always be accomplished by encoding multiple single-channel streams,
it is preferable to take advantage of the redundancy that exists
between channels.
3.4. Teleoperation and Remote Software Services
Teleoperation applications are similar to telepresence, with the
exception that they involve remote physical interactions. For
example, the user may be controlling a robot while receiving real-
time audio feedback from that robot. For these applications, the
delay has to be less than 10 ms. The other requirements of
telepresence (quality, bit-rate, multi-channel) apply to
teleoperation as well. The only exception is that mixing is not an
important issue for teleoperation.
The requirements for remote software services are similar to those of
teleoperation. These applications include remote desktop
applications, remote virtualization, and interactive media
application being rendered remotely (e.g., video games rendered on
central servers). For all these applications, full-band audio with
an algorithmic delay below 10 ms are important.
3.5. In-Game Voice Chat
An increasing number of computer/console games make use of VoIP to
allow players to communicate in real time. The requirements for
gaming are similar to those of conferencing, with the main difference
being that narrowband compatibility is not necessary. While for most
applications a codec delay up to 30 ms is acceptable, a low-delay (<
10 ms) option is highly desirable, especially for games with rapid
interactions. The ability to use variable bit-rate (VBR) (with a
maximum allowed bit-rate) is also highly desirable because it can
significantly reduce the bandwidth requirement for a game server.
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3.6. Live Distributed Music Performances / Internet Music Lessons
Live music over the Internet requires extremely low end-to-end delay
and is one of the most demanding applications for interactive audio
transmission. It has been observed that for most scenarios, total
end-to-end delays up to 25 ms could be tolerated by musicians, with
the absolute limit (where none of the scenarios are possible) being
around 50 ms [carot09]. In order to achieve this low delay on the
Internet -- either in the same city or in a nearby city -- the
network propagation time must be taken into account. When also
subtracting the delay of the audio buffer, jitter buffer, and
acoustic path, that leaves around 2 ms to 10 ms for the total delay
of the codec. Considering the speed of light in fiber, every 1 ms
reduction in the codec delay increases the range over which
synchronization is possible by approximately 200 km.
Acoustic echo is expected to be an even more important issue for
network music than it is in conferencing, especially considering that
the music quality requirements essentially forbid the use of a "non-
linear processor" (NLP) with AEC. This is another reason why very
low delay is essential.
Considering that the application is music, the full audio bandwidth
(44.1 or 48 kHz sampling rate) must be transmitted with a bit-rate
that is sufficient to provide near-transparent to transparent
quality. With the current audio coding technology, this corresponds
to approximately 64 kbit/s to 128 kbit/s per channel. As for
telepresence, support for two or more channels is often desired, so
it would be useful for a codec to be able to take advantage of the
redundancy that is often present between audio channels.
3.7. Delay-Tolerant Networking or Push-to-Talk Services
Internet transmissions are subjected to interruptions of connectivity
that severely disturb a phone call. This may happen in cases of
route changes, handovers, slow fading, or device failures. To
overcome this distortion, the phone call can be halted and resumed
after the connectivity has been reestablished again.
Also, if transmission capacity is lower than the minimal coding rate,
switching to a push-to-talk mode still allows for effective
communication. In this situation, voice is transmitted at slower-
than-real-time bit-rate and conversations are interrupted until the
speech has been transmitted.
These modes require interrupting the audio playout and continuing
after a pause of arbitrary duration.
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3.8. Other Applications
The above list is by no means a complete list of all applications
involving interactive audio transmission on the Internet. However,
it is believed that meeting the needs of all these different
applications should be sufficient to ensure that the needs of other
applications not listed will also be met.
4. Constraints Imposed by the Internet on the Codec
Packet losses are inevitable on the Internet, and dealing with them
is one of the most fundamental requirements for an Internet audio
codec. While any audio codec can be combined with a good packet-loss
concealment (PLC) algorithm, the important aspect is what happens on
the first packets received _after_ the loss. More specifically, this
means that:
o it should be possible to interpret the contents of any received
packet, irrespective of previous losses as specified in BCP 36
[PAYLOADS]; and
o the decoder should re-synchronize as quickly as possible (i.e.,
the output should quickly converge to the output that would have
been obtained if no loss had occurred).
The constraint of being able to decode any packet implies the
following considerations for an audio codec:
o The size of a compressed frame must be kept smaller than the MTU
to avoid fragmentation;
o The interpretation of any parameter encoded in the bit-stream must
not depend on information contained in other packets. For
example, it is not acceptable for a codec to allow signaling a
mode change in one packet and assume that subsequent frames will
be decoded according to that mode.
Although the interpretation of parameters cannot depend on other
packets, it is still reasonable to use some amount of prediction
across frames, provided that the predictors can resynchronize quickly
in case of a lost packet. In this case, it is important to use the
best compromise between the gain in coding efficiency and the loss in
packet loss robustness due to the use of inter-frame prediction. It
is a desirable property for the codec to allow some real-time control
of that trade-off, so that it can take advantage of more prediction
when the loss rate is small, while being more robust to losses when
the loss rate is high.
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To improve the robustness to packet loss, it would be desirable for
the codec to allow an adaptive (data- and network-dependent) amount
of side information to help improve audio quality when losses occur.
For example, side information may include the retransmission of
certain parameters encoded in the previous frame(s).
To ensure freedom of implementation, decoder-side-only error
concealment does not need to be specified, although a functional PLC
algorithm is desirable as part of the codec reference implementation.
Obviously, any information signaled in the bit-stream intended to aid
PLC needs to be specified.
Another important property of the Internet is that it is mostly a
best-effort network, with no guaranteed bandwidth. This means that
the codec has to be able to vary its output bit-rate dynamically (in
real time), without requiring an out-of-band signaling mechanism, and
without causing audible artifacts at the bit-rate change boundaries.
Additional desirable features are:
o Having the possibility to use smooth bit-rate changes with one
byte/frame resolution;
o Making it possible for a codec to adapt its bit-rate based on the
source signal being encoded (source-controlled VBR) to maximize
the quality for a certain _average_ bit-rate.
Because the Internet transmits data in bytes, a codec should produce
compressed data in integer numbers of bytes. In general, the codec
design should take into consideration explicit congestion
notification (ECN) and may include features that would improve the
quality of an ECN implementation.
The IETF has defined a set of application-layer protocols to be used
for transmitting real-time transport of multimedia data, including
voice. Thus, it is important for the resulting codec to be easy to
use with these protocols. For example, it must be possible to create
an [RTP] payload format that conforms to BCP 36 [PAYLOADS]. If any
codec parameters need to be negotiated between end-points, the
negotiation should be as easy as possible to carry over session
initiation protocol (SIP) [RFC3261]/ session description protocol
(SDP) [RFC4566] or alternatively over extensible messaging and
presence protocol (XMPP) [RFC6120] / Jingle [XEP-0167].
5. Detailed Basic Requirements
This section summarizes all the constraints imposed by the target
applications and by the Internet into a set of actual requirements
for codec development.
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5.1. Operating Space
The operating space for the target applications can be divided in
terms of delay: most applications require a "medium delay" (20-30
ms), while a few require a "very low delay" (< 10 ms). It makes
sense to divide the space based on delay because lowering the delay
has a cost in terms of quality versus bit-rate.
For medium delay, the resulting codec must be able to efficiently
operate within the following range of bit-rates (per channel):
o Narrowband: 8 kbit/s to 16 kbit/s
o Wideband: 12 to 32 kbit/s
o Super-wideband: 24 to 64 kbit/s
o Full-band: 32 to 80 kbit/s
Obviously, a lower-delay codec that can operate in the above range is
also acceptable.
For very low delay, the resulting codec will need to operate within
the following range of bit-rates (per channel):
o Super-wideband: 32 to 80 kbit/s
o Full-band: 48 to 128 kbit/s
o (Narrowband and wideband not required)
5.2. Quality and Bit-Rate
The quality of a codec is directly linked to the bit-rate, so these
two must be considered jointly. When comparing the bit-rate of
codecs, the overhead of IP/UDP/RTP headers should not be considered,
but any additional bits required in the RTP payload format, after the
header (e.g., required signaling), should be considered. In terms of
quality versus bit-rate, the codec to be developed must be better
than the following codecs, that are generally considered royalty-
free:
o For narrowband: Speex (NB) [Speex], and internet low bit-rate
codec (iLBC)(*) [RFC3951]
o For wideband: Speex (WB) [Speex], G.722.1(*) [ITU.G722.1]
o For super-wideband/fullband: G.722.1C(*) [ITU.G722.1]
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The codecs marked with (*) have additional licensing restrictions,
but the codec to be developed should still not perform significantly
worse. In addition to the quality targets listed above, a desirable
objective is for the codec quality to be no worse than Adaptive
Multi-Rate (AMR-NB) and Adaptive Multi-Rate Wideband (AMR-WB).
Quality should be measured for multiple languages, including tonal
languages. The case of multiple simultaneous voices (as sometimes
happens in conferencing) should be evaluated as well.
The comparison with the above codecs assumes that the codecs being
compared have similar delay characteristics. The bit-rate required,
for a certain level of quality, may be higher than the referenced
codecs in cases where a much lower delay is required. In that case,
the increase in bit-rate must be less than the ratio between the
delays.
It is desirable for the codecs to support source-controlled variable
bit-rate (VBR) to take advantage of different inputs, that require a
different bit-rate, to achieve the same quality. However, it should
still be possible to use the codec at a truly constant bit-rate to
ensure that no information leak is possible when using an encrypted
channel.
5.3. Packet-Loss Robustness
Robustness to packet loss is a very important aspect of any codec to
be used on the Internet. Codecs must maintain acceptable quality at
loss rates up to 5% and maintain good intelligibility up to 15% loss
rate. At any sampling rate, bit-rate, and packet-loss rate, the
quality must be no less than the quality obtained with the Speex
codec or the Global System for Mobile Communications - Full Rate
(GSM-FR) codec in the same conditions. The actual packet-loss
"patterns" to be used in testing must be obtained from real packet-
loss traces collected on the Internet, rather than from loss models.
These traces should be representative of the typical environments in
which the applications of Section 3 operate. For example, traces
related to VoIP calls should consider the loss patterns observed for
typical home broadband and corporate connections.
5.4. Computational Resources
The resulting codec should be implementable on a wide range of
devices, so there should be a fixed-point implementation or at least
assurance that a reasonable fixed-point is possible. The
computational resources figures listed below are meant to be upper
bounds. Even below these bounds, resources should still be
minimized. Any proposed increase in computational resources
consumption (e.g., to increase quality) should be carefully evaluated
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even if the resulting resource consumption is below the upper bound.
Having variable complexity would be useful (but not required) in
achieving that goal as it would allow trading quality/bit-rate for
lower complexity.
The computational requirements for real-time encoding and decoding of
a mono signal on one core of a recent x86 CPU (as measured with the
Unix "time" utility or equivalent) are as follows:
o Narrowband: 40 megahertz (MHz) (2% of a 2 gigahertz (GHz) CPU
core)
o Wideband: 80 MHz (4% of a 2 GHz CPU core)
o Super-wideband/fullband: 200 MHz (10% of a 2 GHz CPU core)
It is desirable that the MHz values listed above also be achievable
on fixed-point digital signal processors that are capable of single-
cycle multiply-accumulate operations (16x16 multiplication
accumulated into 32 bits).
For applications that require mixing (e.g., conferencing), it should
be possible to estimate the energy and/or the voice activity status
of the decoded signal with less than 10% of the complexity figures
listed above.
It is the intent to maximize the range of devices on which a codec
can be implemented. Therefore, the reference implementation must not
depend on special hardware features or instructions to be present in
order to meet the complexity requirement. However, it may be
desirable to take advantage of such hardware when available, (e.g.,
hardware accelerators for operations like Fast Fourier Transforms
(FFT) and convolutions). A codec should also minimize the use of
saturating arithmetic so as to be implementable on architectures that
do not provide hardware saturation (e.g., ARMv4).
The combined codec size and data read-only memory (ROM) should be
small enough not to cause significant implementation problems on
typical embedded devices. The codec context/state size required
should be no more than 2*R*C bytes in floating-point, where R is the
sampling rate and C is the number of channels. For fixed-point, that
size should be less than R*C. The scratch space required should also
be less than 2*R*C bytes for floating point or less than R*C bytes
for fixed-point.
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6. Additional Considerations
There are additional features or characteristics that may be
desirable under some circumstances, but should not be part of the
strict requirements. The benefit of meeting these considerations
should be weighted against the associated cost.
6.1. Low-Complexity Audio Mixing
In many applications that require a mixing server (e.g.,
conferencing, games), it is important to minimize the computational
cost of the mixing. As much as possible, it should be possible to
perform the mixing with fewer computations than it would take to
decode all the streams, mix them, and re-encode the result.
Properties that reduce the complexity of the mixing process include:
o The ability to derive sufficient parameters, such as loudness
and/or spectral envelope, for estimating voice activity of a
compressed frame without fully decoding that frame;
o The ability to mix the streams in an intermediate representation
(e.g., transform domain), rather than having to fully decode the
signals before the mixing;
o The use of bit-stream layers (Section 6.3) by aggregating a small
number of active streams at lower quality.
For conferencing applications, the total complexity of the decoding,
voice activity detection (VAD), and mixing should be considered when
evaluating proposals.
6.2. Encoder Side Potential for Improvement
In many codecs, it is possible to improve the quality by improving
the encoder without breaking compatibility (i.e., without changing
the decoder). Potential for improvement varies from one codec to
another. It is generally low for pulse code modulation (PCM) or
adaptive differential pulse code modulation (ADPCM) codecs and higher
for perceptual transform codecs. All things being equal, being able
to improve a codec after the bit-stream is a desirable property.
However, this should not be done at the expense of quality in the
reference encoder. Other potential improvements include signal-
adaptive frame size selection and improved discontinuous transmission
(DTX) algorithms that take advantage of predicting the decoder sides
packet loss concealment (PLC) algorithms.
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6.3. Layered Bit-Stream
A layered codec makes it possible to transmit only a certain subset
of the bits and still obtain a valid bit-stream with a quality that
is equivalent to the quality that would be obtained from encoding at
the corresponding rate. While this is not a necessary feature for
most applications, it can be desirable for cases where a "mixing
server" needs to handle a large number of streams with limited
computational resources.
6.4. Partial Redundancy
One possible way of increasing robustness to packet loss is to
include partial redundancy within packets. This can be achieved
either by including the base layer of the previous frame (for a
layered codec) or by transmitting other parameters from the previous
frame(s) to assist the PLC algorithm in case of loss. The ability to
include partial redundancy for high-loss scenarios is desirable,
provided that the feature can be dynamically turned on or off (so
that no bandwidth is wasted in case of loss-free transmission).
6.5. Stereo Support
It is highly desirable for the codec to have stereo support. At a
minimum, the codec should be able to encode two channels
independently without causing significant stereo image artifacts. It
is also desirable for the codec to take advantage of the inter-
channel redundancy in stereo audio to reduce the bit-rate (for an
equivalent quality) of stereo audio compared to coding channels
independently.
6.6. Bit Error Robustness
The vast majority of Internet-based applications do not need to be
robust to bit errors because packets either arrive unaltered or do
not arrive at all. Therefore, the emphasis should be on packet-loss
robustness and packet-loss concealment. That being said, often, the
extra robustness to bit errors can be achieved at no cost at all
(i.e., no increase in size, complexity, or bit-rate; no decrease in
quality, or packet-loss robustness, etc.). In those cases, it is
useful to make a change that increases the robustness to bit errors.
This can be useful for applications that use UDP Lite transmission
(e.g., over a wireless LAN). Robustness to packet loss should
*never* be sacrificed to achieve higher bit error robustness.
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6.7. Time Stretching and Shortening
When adaptive jitter buffers are used, it is often necessary to
stretch or shorten the audio signal to allow changes in buffering.
While this operation can be performed directly on the decoder's
output, it is often more computationally efficient to stretch or
shorten the signal directly within the decoder. It is desirable for
the reference implementation to provide a time stretching/shortening
implementation, although it should not be normative.
6.8. Input Robustness
The systems providing input to the encoder and receiving output from
the decoder may be far from ideal in actual use. Input and output
audio streams may be corrupted by compounding non-linear artifacts
from analog hardware and digital processing. The codecs to be
developed should be tested to ensure that they degrade gracefully
under adverse audio input conditions. Types of digital corruption
that may be tested include tandeming, transcoding, low-quality
resampling, and digital clipping. Types of analog corruption that
may be tested include microphones with substantial background noise,
analog clipping, and loudspeaker distortion. No specific end-to-end
quality requirements are mandated for use with the proposed codec.
It is advisable, however, that several typical in situ environments/
processing chains be specified for the purpose of benchmarking end-
to-end quality with the proposed codec.
6.9. Support of Audio Forensics
Emergency calls can be analyzed using audio forensics if the context
and situation of the caller has to be identified. Thus, it is
important to transmit not only the voice of the callers well, but
also to transmit background noise at high quality. In these
situations, sounds or noises of low volume should also not be
compressed or dropped. Therefore, the encoder must allow DTX to be
disabled when required (e.g., for emergency calls).
6.10. Legacy Compatibility
In order to create the best possible codec for the Internet, there is
no requirement for compatibility with legacy Internet codecs.
7. Security Considerations
Although this document itself does not have security considerations,
this section describes the security requirements for the codec.
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As for any protocol to be used over the Internet, security is a very
important aspect to consider. This goes beyond the obvious
considerations of preventing buffer overflows and similar attacks
that can lead to denial-of-service (DoS) or remote code execution.
One very important security aspect is to make sure that the decoders
have a bounded and reasonable worst-case complexity. This prevents
an attacker from causing a DoS by sending packets that are specially
crafted to take a very long (or infinite) time to decode.
A more subtle aspect is the information leak that can occur when the
codec is used over an encrypted channel (e.g., [SRTP]). For example,
it was suggested [wright08] [white11] that use of source-controlled
VBR may reveal some information about a conversation through the size
of the compressed packets. Therefore, it should be possible to use
the codec at a truly constant bit-rate, if needed.
8. Acknowledgments
We would like to thank all the people who contributed directly or
indirectly to this document, including Slava Borilin, Christopher
Montgomery, Raymond (Juin-Hwey) Chen, Jason Fischl, Gregory Maxwell,
Alan Duric, Jonathan Christensen, Julian Spittka, Michael Knappe,
Christian Hoene, and Henry Sinnreich. We would also like to thank
Cullen Jennings, Jonathan Rosenberg, and Gregory Lebovitz for their
advice.
9. Informative References
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC6120] Saint-Andre, P., "Extensible Messaging and Presence
Protocol (XMPP): Core", RFC 6120, March 2011.
[XEP-0167] Ludwig, S., Saint-Andre, P., Egan, S., McQueen, R., and
D. Cionoiu, "Jingle RTP Sessions", XSF XEP 0167,
December 2009.
[RFC3951] Andersen, S., Duric, A., Astrom, H., Hagen, R., Kleijn,
W., and J. Linden, "Internet Low Bit Rate Codec (iLBC)",
RFC 3951, December 2004.
Valin & Vos Informational [Page 15]
RFC 6366 Audio Codec Requirements August 2011
[ITU.G722.1] International Telecommunications Union, "Low-complexity
coding at 24 and 32 kbit/s for hands-free operation in
systems with low frame loss", ITU-T Recommendation
G.722.1, May 2005.
[Speex] Xiph.Org Foundation, "Speex: http://www.speex.org/",
2003.
[carot09] Carot, A., Werner, C., and T. Fischinger, "Towards a
Comprehensive Cognitive Analysis of Delay-Influenced
Rhythmical Interaction:
http://www.carot.de/icmc2009.pdf", 2009.
[PAYLOADS] Handley, M. and C. Perkins, "Guidelines for Writers of
RTP Payload Format Specifications", BCP 36, RFC 2736,
December 1999.
[RTP] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[SRTP] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and
K. Norrman, "The Secure Real-time Transport Protocol
(SRTP)", RFC 3711, March 2004.
[wright08] Wright, C., Ballard, L., Coull, S., Monrose, F., and G.
Masson, "Spot me if you can: Uncovering spoken phrases
in encrypted VoIP conversations:
http://www.cs.jhu.edu/~cwright/oakland08.pdf", 2008.
[white11] White, A., Matthews, A., Snow, K., and F. Monrose,
"Phonotactic Reconstruction of Encrypted VoIP
Conversations: Hookt on fon-iks
http://www.cs.unc.edu/~fabian/papers/foniks-oak11.pdf",
2011.
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RFC 6366 Audio Codec Requirements August 2011
Authors' Addresses
Jean-Marc Valin
Mozilla
650 Castro Street
Mountain View, CA 94041
USA
EMail: jmvalin@jmvalin.ca
Koen Vos
Skype Technologies, S.A.
Stadsgarden 6
Stockholm, 11645
Sweden
EMail: koen.vos@skype.net
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