Internet DRAFT - draft-hoene-avt-acs-requirements
draft-hoene-avt-acs-requirements
AVT Christian Hoene
Internet Draft University of Tuebingen
Intended status: Informational August 17, 2009
Expires: February 2010
Requirements of an Audio Communication System (ACS)
draft-hoene-avt-acs-requirements-00.txt
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Abstract
This document describes the requirements of an audio communication
system (ACS) for acoustic content, especially speech and music. The
ACS consists of all components above the IP layer and below a digital
PCM audio interface. These include codec, jitter buffer, and
transport.
The goal of the ACS is to provide a bidirectional acoustic
communication between any two Internet hosts at a good quality,
constrained only by the available resources at the hosts and the
characteristics of the transmission path between both hosts.
The intention of the document is to provide the requirements for a
codec that is solely intended for the Internet, to provide the
requirements for the codec's payload specification, and to define the
requirements on the transport protocol.
Table of Contents
1. Introduction...................................................3
1.1. Basics Architectural Guidelines of the Public Internet....4
1.2. Problem Statement.........................................5
2. Usage Scenarios................................................7
2.1. Scenario 1: Person-to-person calls (VoIP).................8
2.2. Scenario 2: High quality interactive audio transmissions
(AoIP).........................................................8
2.3. Scenario 3: Ensembles performing over a network (MMoIP)...9
2.4. Scenario 3: Push-to-talk like service (PTT)...............9
3. High-Level Requirements.......................................10
3.1. Low cost and licensing free..............................10
3.2. Reliable on the Internet.................................11
3.3. Quality..................................................11
4. Technical Requirements........................................12
4.1. Audio content............................................12
4.2. Quality..................................................12
4.3. Reliability and congestion control.......................12
4.4. Coding bit rate..........................................13
4.5. Sampling rate............................................13
4.6. Complexity...............................................13
4.7. Latency..................................................14
4.8. Packet rate..............................................14
4.9. Packet loss resilience...................................15
4.10. Frame erasure concealment...............................15
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4.11. Jitter compensation and playout buffer..................15
4.12. Playout adjustments.....................................16
4.13. Concealment of mode switches............................16
4.14. Extrapolation...........................................16
4.15. Interpolation...........................................17
4.16. DTX.....................................................17
4.17. Testing.................................................17
4.18. Licensing and source code...............................17
4.19. Versioning and software updates.........................18
4.20. RFC Type................................................18
4.21. Side channel............................................18
4.22. Layered coding..........................................18
4.23. Interoperability with PSTN..............................19
4.24. Conferencing and speech recognition.....................19
4.25. Self-testing support....................................19
4.26. Self-awareness..........................................19
5. Out of scope..................................................19
5.1. Multichannel.............................................19
5.2. Repacketization..........................................19
5.3. Support for circuit-switched transmissions...............19
5.4. Support of packet networks other than the Internet.......20
5.5. Support of streaming.....................................20
5.6. Random packet losses.....................................20
5.7. Packet loss differentiation..............................20
5.8. Robustness against bit errors............................20
5.9. IRS and other kind of bandwidth filters..................20
5.10. Support of voice band data, fax and DTMF................20
5.11. Idle noise..............................................21
5.12. Tandem coding...........................................21
5.13. FEC.....................................................21
6. Security Considerations.......................................21
7. IANA Considerations...........................................21
8. References....................................................21
8.1. Normative References.....................................21
8.2. Informative References...................................22
9. Acknowledgments...............................................23
1. Introduction
This document is based mainly on the discussions on the Codec BOF
mailing list, which took place in 2009. It also based on the internal
requirement documents of ITU-T G.718 [SG16 314-WP3], on the ITU-T
G.719 standard, on the 3GPP document [TS26.114-830], and on existing
IETF codec drafts.
It is intended as basis of a requirement document that should lead to
the design of an audio codec for the Internet. However, this document
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address the requirements of the entire system not only of a single
component because we want to ensure that the system as a hole works
well not only some parts of it.
We introduce the term audio communication system to describe the
parts of an IP based telephone which are care for the bidirectional
transmission of acoustic content between two Internet hosts. These
include the encoder, the payload encapsulation, guide lines on how to
use transport protocols (RTP, UDP, TCP, DCCP), the playout buffer,
the decoder, the concealment of packet loss, time adjustments,
changes of encoding parameter, and various mechanisms to manage,
control and monitor the acoustic transmission.
The ACS is intended mainly for the use on the public Internet and
should be as easily distributable as most other Internet protocols
that run on virtually all kind of devices and on all kind of
communication links. Also, the ACS shall be affordable by all humans
that have Internet access. If possible, it should be royalty free and
available as open-source software. If these requirements are given,
then the ACS can fulfill its goal of providing acoustic transmission
between _any_ two Internet hosts.
1.1. Basics Architectural Guidelines of the Public Internet
The ACS is intended for the public Internet and follows similar
architectural design guidelines as those which are valid for other
Internet protocols, too. These include:
o End-to-end semantics saying that transport protocol units are
transmitted from one end (an Internet host) to the other end
without any intermediate changes.
o Network neutrality.
o Best effort service that try to transmit packets as good as
possible but that cannot guaranty any minimal transmission
bandwidth or maximal transmission delay. Instead one has to cope
with any end-to-end transmission quality that is provided.
o Congestion control to prevent congestion collapse of the Internet
(such as TCP or DCCP). Typically, TCP controls the number of
packets that are sent during periods of congestion. Thus, one has
to consider that the number of packets per second might be an
important constraining limitation and not only the bits per
second.
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o Internet protocols are scalable to wide degree. They work on links
having a very low bandwidth (in the order of bits per second) and
with very high bandwidth (in the order of gigabits per second).
The transmission latency can range from microsecond up to seconds.
Also, the Internet hosts might have very low processing and memory
capabilities (such as an 8-bit micro controller). However, even
then they can communicate with any other hosts. Flow control (such
as in TCP) is used to cope with hosts that have limited resources.
o Functions to help monitoring the communication (such as the
features provided ICMP)
o The most important Internet protocols can be used without paying
royalties.
o The public Internet allows global communication between any two
hosts connected to the public Internet. Typically, the user only
has to pay for getting access to the public Internet not for the
distance that the IP packets have to travel.
o Internet standards should be as simple as possible (but no
simpler).
1.2. Problem Statement
The ACS should enable an acoustic communication between any two
Internet hosts considering the features of the Internet as described
above. We see the need for designing the ACS because we see the
following weaknesses in the existing codec and VoIP designs.
o Many standardized speech and audio codecs require the payment of
royalty fees. Only codecs such as G.711, G.722, G722.1, and
G.722.1C that have mediocre performances can be used license free.
Thus, one cannot ensure that a good codec can be afforded by all
owners of all Internet hosts.
o All known codecs have a small operational range, in addition they
do not adapt to a wide range of bandwidth. For example, AMR
support bit rates between 4.75 and 12.2 kbps and ITU G.719 support
rates between 32 kbps and 128 kbps.
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o An acoustic communication at superb transmission quality is not
supported. Especially, if the latency is very low and the
bandwidth is very high, we do not have a standardized codec that
support hifi quality at ultra low delays. Only the SBC audio codec
standardized by Bluetooth SIG [A2DPV10] can be considered for the
usage scenario.
Ultra-low delay transmissions at hifi quality are especially
useful for distributed ensemble performances or distributed
choruses.
o Similar, if the transmission quality is very bad, no standardized
audio codec supports a grateful degradation. If the loss rate
becomes too high then all speech and audio codecs become useless.
However, in those cases one can use half-duplex, push-to-talk like
transmission of short audio segments that would still allow a very
slow communication at very low bitrates.
o Frequently, a PSTN call needs to be transcoded. Transcoding
reduces the speech quality and increases latency. Thus, most
codecs are designed to work well in conditions of transcoding.
However, in case end-to-end IP transmission, the need for
transcoding vanishes. It might only be needed for teleconferencing
applications or for connecting to the PSTN network.
o The quality of a PSTN call has hardly increased during the last
decade. Often, it is even worse because of IP based
interconnections and support of cellular networks. Even those
support of wideband speech transmission system have been
developed, the lack of the willingness of users to pay more has
limited the introduction of wideband speech. Also on the Internet
we do not expect users to pay more for high quality phone calls.
However, we believe that they will be delighted if they can
communicate at nearly perfect quality.
o Neither a single standardized codec nor its RTP payload RFC
specifies how to cope with time varying bandwidth and latency nor
this is considered as required feature. This hinders the wide
spread use of an adaptive coding mode selection and thus reduces
the quality of many Internet phone calls.
o Not a single standardized codec supports varying complexities to
support devices with low resources.
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o Standardized codecs do not support any functionality for self-
observation and self-monitoring. Also, they do not provide
information about how well they encoded and decoded the audio
content under a given set of coding parameters and packet loss
rates. However, this information is important for the transport in
order to rightly adapt the codec's transmission parameters.
o Packet losses occur in the Internet, the transmission time of
packets and the playout time varies and the coding mode is changed
in response to changed available transmission bandwidths. All
these things cause the audio stream to be temporally distorted.
The codec shall support concealment algorithms to limit the
perceptual distortion. However, none existing standardized codec
support the concealment of the adjustment of the playout time.
Also, standardized PLC work on extrapolation of previous audio
segments and do not support the interpolation. Lastly, often one
cannot distinguish between delaying the playout time and packet
loss because the missing packet might still arrive. Thus, an
algorithm that uses the same extrapolation for packet loss
concealment and time stretching might be beneficial.
o None of the standardized interactive speech and audio codec
supports mechanisms to decrease the packet rate. Usually, packet
rates are reduced by putting multiple speech frames into on RTP
packet. However, the codecs do not take advantage of the high
algorithmic delay that can be utilized then. Thus, they work less
efficient in situations of congestion.
2. Usage Scenarios
The ACS should be optimized towards real-time communications over the
Internet. It should support applications like collaborative network
music performance, high-quality teleconferencing, wireless audio
equipment, low-delay links for broadcast applications, network sound
servers for using multimedia applications remotely, telepresence
(enterprise) and the digital living room (consumer), and other.
The ACS shall be general enough to support multiple and quite diverse
network conditions. For example, if network latency is low and
bandwidth is plenty, it can be used for quasi-simultaneous music
transmissions allowing distributed ensemble performances. It is also
applicable interactive hifi quality audio transmission. If the
network connection worsens, the transmission quality degrades to
(wide-band) interactive speech transmission. As a last resort, it
emulates a high-delay, half-duplex push-to-talk like communication
service.
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In the following, we enlist four main scenarios and describe their
quality requirements.
2.1. Scenario 1: Person-to-person calls (VoIP)
The classic scenario is that of the phone usage to which we will
refer in this document as Voice over IP (VoIP). Human speech is
transmitted interactively between two Internet hosts. Typically,
beside speech some background noise is present, too.
The quality of a telephone call is traditionally judged with
subjective tests such as those describe in [ITU-T P.800]. The ACR
scale used in MOS-LQS sometimes might not be very suitable for high
quality, then - for example - the MUSHRA [ITU-T BS.1534-1] rating can
be applied.
A telephone call is considered good if it has a maximal mouth-to-ear
delay of 150ms [ITU-T G.107] and a speech quality of MOS-LQS 4 or
above. However, interhuman communication is still possible if the
delay is much larger.
This scenario does not include the use case of using a VoIP-PSTN
gateway to connected to legacy telephone systems. In those cases, the
gateway would make an audio conversion from broadband Internet voice
to the frugal 1930's 3.1 kHz audio bandwidth. Interconnections to the
PSTN will most likely stick with its legacy codecs to avoid
transcoding.
2.2. Scenario 2: High quality interactive audio transmissions (AoIP)
In this first scenario we consider a telephone call having a very
good audio quality at modest acoustic one-way latencies ranging from
50 and 150 ms [ITU-T G.107], so that music can be listened over the
telephone while two persons talk interactively.
The Absolute Category Rating (ACR) (refer to ITU-T P.800) can be
used, too. However, it might be more efficient to measure quality
with the MUSHRA tests given in [ITU-T BS.1534-1], which is intended
for intermediate audio qualities.
Also, for today's teleconferencing and videoconferencing systems
there is a strong and increasing demand for audio coding providing
the full human auditory bandwidth of 20 Hz to 20 kHz. This rising
demand for high quality audio is due to the following:
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o Conferencing systems are increasingly used for more elaborate
presentations, often including music and sound effects which
occupy a wider audio bandwidth than that of speech. For example,
Web conferences such as WebEx, GoToMeeting, Adobe Acrobat Connect
are based on an IP based transmission and benefit from a IP
optimized ACS.
o The new "Telepresence" video conferencing systems, providing High
Definition video and audio quality to the user, are giving the
experience of being in the same room by introducing high quality
media delivery (such as from Cisco).
o The emerging Digital Living Rooms will likely be interconnected
and might require a constant acoustic transmission at high
qualities.
2.3. Scenario 3: Ensembles performing over a network (MMoIP)
In some usage scenarios, users want to act simultaneously and not
just interactively. For example, if persons sing in a chorus, if
musicians jam, or if e-sportsmen play computer games in a team
together they need to acoustically communicate. We call it the Make
Music Over IP (MMoIP) scenario.
In this scenario, the latency requirements are much harder than for
interactive usages. For example, if two musicians are placed more
than 10 meters apart, they can hardly keep synchronized. Empirical
studies [Gurevich2004] have shown that if ensembles playing over
networks, the optimal acoustic latency is around 11.5 ms with
targeted range from 10 to 25 ms.
In addition to the MUSHRA tests, the recommendation [ITU-R BS.1116]
can be used for audio transmissions that just have minor impairments.
2.4. Scenario 4: Push-to-talk like service (PTT)
In spite of the development of broadband access (xDSL), a lot of
users would only have service access via PSTN modems or mobile links.
Also, on these links the available bandwidth might be shared among
multiple flows and is subjected to congestion. Then, even low coding
rates at about 8 kbps are too high.
If transmission capacity hardly exists, one still can degrade the
quality of a telephone call to something like a push-to-talk (PTT)
like service having very high latencies. Technically, this scenario
takes advantage of bandwidth gains due to disruptive transmission
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(DTX) modes and very large packets containing multiple speech frames
causing a very low packetization overhead.
The quality requirements of a push to talk like service have been
hardly studied. The OMA lists as a requirement of a Push To Talk over
Cellular service a transmission delay of 1.6 s and a MOS values of
above 3.0 that typically should be kept [OMAPoCReq]. However, as long
as an understandable transmission of speech is possible, the delay
can be even higher. For example, [OMAPoCReq] allows a delay of
typically up 4s for the first talk-burst.
Also, [OMAPoCReq] describes a maximum duration of speaking. If a
participant speaking reaches the time limit, the participant's right-
to-speak shall be automatically revoked.
If the quality of a telephone call is very low, then instead of
listening-only speech quality the degree of understandability can be
chosen as performance metric. For example, objective tests of the
understandability use automatic speech recognition (ASR) systems and
measure the amount of correctly detected words.
In any case, the participant shall be informed about the quality of
connection, the presence of high delays, the half-duplex style of
communication, and its (limited) right-to-speak. For example this can
be achieved by a simulated talker echo.
3. High-Level Requirements
Based on the four scenarios, we list the following high-level
requirements that the ACS should fulfill.
3.1. Low cost and licensing free
The codec shall be affordable by all humans having Internet access.
Thus, one of the key requirements is patent/licensing free
technology. However, it cannot be seen as "legally binding
requirement" but rather as a desired working goal. Typically, one
cannot verify 100% whether a codec is totally free of unknown IPRs.
Some patents may be overlooked. It can also be assured that the known
IPRs are "license-free" and "free from the need to sign licensing
agreement(s) before use" (The ability for any user to get the codec
and use it without signing any paperwork).
If one is practicing potentially patented technologies, there is no
real mechanism to protect oneself from a patent troll at claims
license fee for a standardized ACS. We have to assume that there is a
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certain probability that the designed ACS is covered by patents what
the IETF is not aware of. Thus, one has to define proper procedures
on how to cope with IPR claims even if the ACS is already
standardized.
Because of the lack of financial income, the codecs design, testing
and standardization process must be cost effective, too. A cheap
approach is needed to characterize the ACS, which might include tests
having volunteer participants. For example, codecs can be provided to
thousands of users in public to test them. Also, potential
performance comparisons must not be as precise and proven as beyond
any doubts because nobody wins or loses IPR fees if one solution wins
or fails.
3.2. Reliable on the Internet
The ACS must be optimized towards acoustic real-time communications
over the Internet, and must have the flexibility to adjust to the
environment it operates in. Based on the quality of the end-to-end
speech packet transmission, the codec should adapt its quality and
delay to achieve an optimal benefit for the user.
As most Internet transport, it should be used with a wide range of
condition allowing a high reliability regardless the networking
condition. The reliability of the audio transmission should be high,
even in cases of low and varying bandwidth. This implies that the
codec is used on top of a transport protocol that implements a
congestion control algorithm and that the ACS adapts to changes of
available bandwidth. For example, if the available transmission
bandwidth is too low to allow the codec to transmit audio at a high
quality, the application can lower the sampling, bit or frame rate of
the stream at the cost of higher algorithmic delay or a degraded
audio quality.
3.3. Quality
The ACS must provide a quality/bitrate trade-off that is competitive
with other state-of-the-art codecs. Also, the codec must have a very
low algorithmic delay so that it can support the typical requirements
of its users.
The speech and audio quality of the ACS should not be significantly
worse than existing standardized codecs, if measures on the ACR
scale.
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4. Technical Requirements
4.1. Audio content
At all bitrates the ACS must deliver speech in any language at good
quality. The ACS must be tested for different speakers and at least
with two languages and should support tonal languages as well.
Frequently, speech needs to be transmitted not only without
background noise but also at conditions including car, office and
street noise. Background signals shall be considered not as the noise
but as a part of the signals that convey information. Background
signal can include background music at a SNR of 25 dB, office noise
at a SNR of 20 dB, car noise at a SNR of 15 dB, babble Noise at a SNR
of 25 dB, interfering talker at a SNR of 15 dB and street noise at a
SNR of 20 dB.
At high bitrates the quality must be excellent for any audio signal,
especially music. Stereo is considered as a must. Also, for high
quality audio conferencing, reverberant input signals should be
considered for testing the modes.
The speech and audio signals might have varying loudness. The
transmission shall support a wide range of dynamics. The nominal
input level of -36 dB, -26 dB and -16dB with respect to the
overlapping bandwidth limit (OVL) point (-20 dBm0).
4.2. Quality
At a given operational mode, the ACS must not have perfect quality
and must not perform better than any other standardized codec.
However, considering the most common network conditions, the ACS
shall perform better than any combination of existing codecs most of
the time.
4.3. Reliability and congestion control
The acoustic transmission should be reliable and robust. The ACS
shall be not only robust against packet losses but also for periods
of low bandwidth.
The mean availability of the audio transmissions, calculated over all
users, might be one of the metrics for assessing the performance of
an Internet audio codec.
The ACS should adapt to the current network situation. Also, the
codecs of ACS themselves must be adaptable, because switching among
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multiple codecs is difficult to negotiate and unlikely to work well
in situations of inter-operation.
Responding to congestion is a more complex issue and out of the scope
of this document. However, it shall be defined on how to use existing
congestion control protocols like DCCP and TCP. The ACS shall provide
the mechanisms that congestion control requires from the codec (i.e.
bitrate/framerate adaptability).
Because of the interactive nature of the acoustic transmission, the
bidirectional transmission of audio content can be used for
transmitting the required feedback and implementing a control loop.
As such, it can be considered as a requirement that the acoustic
transmission should be always bidirectional--even if the backward
channel just sends "compressed silence".
4.4. Coding bit rate
The ACS must be capable of running at bitrates below 10 kbps. At low
bitrates it must deliver good quality for clean, noisy or hands-free
speech in any language. At high bitrates the quality must be
excellent for any audio signal, including music. The bitrate must be
adjustable in real-time. The bit rate can go up to 128 kbit/s per
channel or more. The bitrate must be adjustable in real-time and at a
fine granularity.
Variable bit rates depending on the content should be supported.
4.5. Sampling rate
The codec must support multiple sampling rates, ranging from 8 kHz to
full band. Switching between sampling rates must be carried out in
real-time.
4.6. Complexity
The ACS should have a complexity that is adjustable in real-time,
where a higher complexity setting improves the quality/bitrate trade-
off.
As a lower limit, the ACS shall run on hosts that common in
developing countries. These may include OLPC XO-1s or other low-end
(refurbished) computers (refer to Computer Aid International) and
smart phones like those based on Texas Instruments Open Multimedia
Application Platform (OMAP), which include both a host ARM CPU and
one or more DSP.
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On those devices, the ACS must not be capable of running at highest
quality but at least at 8 kHz sampling rate.
4.7. Latency
To maintain a good quality of services requiring interactivity, it is
necessary to maintain the overall delay as low as possible. But the
delay requirement tends to have less importance in applications
involving VoIP, possibly combined with other media and/or in
heterogeneous network environment. A trade-off must be found between
low delays and flexibility (scalability, ability to operate in
various conditions with many types of signals etc.).
In interactive scenarios, the codec should be capable of running with
an algorithmic delay of no more than 30 milliseconds.
For the making music scenario, the algorithmic delay must be between
3 to 9 ms. Still, given the speed of light as the fundamental limit
of speed of information exchange, distributed ensembles can perform
only regionally if latency budget of 25 ms must be kept. Typically,
an optical fiber has a refractive index of 1.46 and thus in an
optical fiber bits travel about 5136 km one-way in 25 ms.
The total codec delay consists of the algorithmic delay and the
processing delay. Algorithmic delay includes the frame size delay
plus any other delays inherent in the algorithm (look-ahead, noise
suppression and error correcting codes for algorithm purposes and any
algorithmic decoding delay). Processing delay is the additional delay
caused by implementation with a finite speed processor.
4.8. Packet rate
The ACS must support a variable and dynamic changeable packet rate.
Putting several frames into one packet is useful for packet grouping,
which in turn is very useful for bandwidth adaptation and network
usage efficiency.
This is because of the fact that a lot of bandwidth is used for
protocol packet headers like those of Ethernet, IP, UDP, and RTP and
thus to overhead at the MAC layer. If even IP header compression is
applied, still many layer 2 protocols introduce an additional
overhead that is not compressed [Hoene2005].
Classically, it is usually specified in the RTP payload
specification, not in the codec specification itself. In general, a
codec can take advantage of a larger frame size. This is especially
true for a transform codec, where a larger frame means better
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frequency resolution. The gain is somewhat smaller time-domain codec
especially for > 20 ms frames. However, in larger packets the inter-
frame dependencies can be adjusted on the fly to choose a trade-off
between bitrate and amount of error propagation. It may even be
possible to just make use of more inter-frame correlation for frames
2...N in a packet of N frames and get most of the benefits it would
get from a larger frame size. Thus, the ACS codec should support
large frame sizes (up to a MTU).
4.9. Packet loss resilience
The codec must be capable of running with little error propagation,
meaning that the decoded signal after one or more packet losses is
close to the decoded signal without packet losses after no more than
two additional packets. The codec must have a packet loss resilience
that is adjustable in real-time, where a lower packet loss resilience
setting improves the quality/bitrate trade-off.
Also, the codec may add inter-frame redundancies to achieve better
loss robustness.
4.10. Frame erasure concealment
The ACS must have a packet loss concealment algorithm. The PLC must
be standardized to know how well the decoder can cope with packet
losses in cases when the transmission parameters must be adjusted.
However, the ACS may implement a PLC that performs better than the
standardized PLC.
The purpose of standardizing the PLC (and the other concealment
algorithms) is to guarantee a certain quality level over a range of
conditions. For good results, a PLC operates on decoder-internal
parameters and states, which requires tight algorithmic integration.
So the PLC is as much part of a decoder as any other decoder module.
The above also applies to time compression/stretching methods for
handling network jitter and other kind of concealment algorithms (as
mentioned below).
4.11. Jitter compensation and playout buffer
The ACS must cope with jitter. It must be able to receive the out of
order de-packetized frames and present them in order for decoder
consumption. It must be able to receive duplicate speech frames and
only present unique speech frames for decoder. It must be able to
handle clock drift between the encoding and decoding end-points.
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The playout buffer should minimize the buffering time at all times
while still conforming to the minimum performance requirements. If
the limit of jitter induced concealment operations cannot be met, it
is always preferred to increase the buffering time in order to avoid
growing jitter induced concealment operations.
4.12. Playout adjustments
The ACS should support time scale modifications especially for jitter
compensations such as time stretching and time shrinking because on
the Internet jitter is the norm not a special case.
Because the operations going on in time scale modification algorithms
are similar as those for the PLC, these operations should be combined
into a single algorithm.
Also, the ACS shall be able to determine a desired length of a time
scale modification (so it can e.g. leave out or add one or more pitch
periods), to keep a 'backup' decoder state of the previous frame or
to add one more frame length of decoding latency - otherwise you can
not compress the voice of the previous packet and for stretching its
suboptimal.
In general, the use of a high-quality time scaling algorithm is
recommended. The amount of scaling should be as low as possible,
scaling should be applied as infrequently as possible, and
oscillating behavior is not allowed.
4.13. Concealment of mode switches
The ACS should also support the concealment of distortions caused by
switching coding modes [Hoene2005]. Also, the negative effect of
switching the coding mode shall be low.
For example, the transmission and coding mode might change several
times (up to 5Hz) per second after getting feedback from the decoder.
4.14. Extrapolation
Sometimes, it is not possible to distinguish between a packet that
arrives too late and packet that is lost and needs to be concealed.
The decision on whether to conceal the loss or whether to conduct
time stretching cannot be made yet. Thus, the ACS should support a
general extrapolation of the audio signal which allows for late
decision on whether to play out a delayed packet or whether to use a
loss concealment operation
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4.15. Interpolation
If a packet n has not arrived but the previous packet n-1 and the
following packet n+1, when the packet n shall be interpolated using
the frame of the previous and following packets.
4.16. DTX
The codec must be capable of using Discontinuous Transmission (DTX)
where packets are sent at a reduced rate when the input signal
contains only background noise.
4.17. Testing
The testing of ACS and the quality characterization shall be
performed with real network profiles such as with [TIA-921] or those
given in the appendix [TS.26114-830], not with fixed set of "average
distributed errors and losses". Later do not clearly reflect the
Internet nature.
Also, test vectors might be provided to check the correctness of the
implementations.
4.18. Licensing and source code
The usage of ACS should not require paying royalties and signing NDA.
At the time of standardization it should be available for royalty
free (RF) and at reasonable and non-discriminatory terms (RAND). The
codec should be available as open source allowing implementation
under BSD, LGPL and/or GPL.
The codec specification description and implementation shall be based
on a bit-exact fixed-point modular ANSI-C code using basic operators
set provided in the ITU-T Software Tool Library to follow. In
addition, an interoperable floating-point implementation can be
provided.
The source code shall be normative because of a number of
reasons. One is ease of implementation (either using the reference
code directly, or being able to use it to validate the ported code).
Another is that it assures that the characterization tests actually
measure the standard's performance. Even if it is not officially
normative, readily available reference code becomes de facto
normative, since most implementers will simply use the code and
ignore the text in the RFC.
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4.19. Versioning and software updates
In order to cope with changes in the bitstream format, which might be
required due to errors in the specification or - more important - due
to newly claimed IPR, it must be possible to update the ACS online.
Also, it must be indicated, which bitstream format is going to be
used.
4.20. RFC Type
It should become a standard, not an experimental RFC.
4.21. Side channel
Congestion control should be must for all Internet applications also
for the ACS. [RFC3550] suggests in Chapter 10 somewhere that the RTP
profile should care for rate adaptation. Thus, the ACS should take
advantage of a feedback loop for variable coding parameter control in
order to allow a wide range of operation and to adapt to the the
current available bandwidth and processing power.
Congestion control per se is outside the review of this group, but
providing the hooks for a congestion-control mechanism to interact
with the codec is quite important. For example, running this codec on
a TFRC-enabled or DCCP RTP stream - TFRC and DCCP need to be able to
adjust (via the application) the bitrate of the codec in order to
implement congestion control and perhaps adjust packetization
periods/packet-rates.
A side channel for adaptation can be added. This would make sense
because in usage scenarios audio is always transmitted in both
directions. Adding a control channel would give a real advantage to
existing codec designs. Alternatively, such as side channel can be
also added with alternative solutions, such as handling that
communication in SIP/SDP and in RTP/RTCP.
4.22. Layered coding
The ACS can support a layered encoding like in G.729.1 and G.718.
Layered coding can be seen as a method for computationally efficient
transcoding. Layered coding make sense in the conferencing
environment as such stripping should be done at the sender after
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encoding. Then, for all receivers the encoding has to be done only
once.
However, for bidirectional transmissions, you do not need layered
encoding as most codecs now are VBR, its enough already to adapt
codec (at the source) to the bandwidth. Also, layered coding comes at
additional cost (about 10% of the coding rate)
4.23. Interoperability with PSTN
The ACS might be developed to be interoperability to existing PSTN
systems. Especially interoperability with 2G and 3G mobile radio
systems is desirable. Also, the interoperability with G.722.2 @ 12,65
kb/s and with G.722 (for DECT devices) are of particular interest.
4.24. Conferencing and speech recognition
A teleconference server should be able to mix the audio signals at
lower complexity than decoding + encoding. The ACS shall be capable
of support automatic speech recognition.
4.25. Self-testing support
ACS should support means of testing the quality of a connection by
feedback loops and quality feedbacks.
4.26. Self-awareness
The ACS should be aware on how well it can transmit acoustic content
at various coding parameters and packet loss rates.
5. Out of scope
5.1. Multichannel
5.1 is worth supporting but that would most likely be through
multiple independent channels/pairs, so that's probably not that much
of an issue.
5.2. Repacketization
The ACS needs not to support repacketization in a network because
this would violate the end-to-end semantic of the Internet.
5.3. Support for circuit-switched transmissions
The ACS needs not to support circuit-switched transmission.
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5.4. Support of packet networks other than the Internet
The ACS needs not to support other packet networks (VoATM, private
networks) beside the Internet.
5.5. Support of streaming
The ACS needs not to support multimedia streaming (e.g. video + audio
involving bit-rate tradeoff), multicast content distribution
(offline/online) and message retrieval systems.
5.6. Random packet losses
The usage of random packet losses to measure the concealment
performance is meaningless because it does not reflect the nature of
the Internet. Thus, the codec needs not be optimized nor tested using
these criteria. Instead, real packet loss and delay traces should be
considered. Also, short and long bursts of packet losses, which occur
during due to handoffs, fast fading, congestion events, and route
changes, should be considered.
5.7. Packet loss differentiation
The ACS cannot assume that the quality of packet transmission changes
one per packet basis. For example, in layered coding the core layers
cannot expect to be less subjected to packet losses than enhancement
layers.
5.8. Robustness against bit errors
The ACS needs not to support bit errors because they are quite seldom
on top of Ethernet. This is especially true as long as UDP-Lite is
not supported widely.
5.9. IRS and other kind of bandwidth filters
The ACS must not consider bandwidth filters like the IRS because they
are based on the traditions of circuit-switched connections.
5.10. Support of voice band data, fax and DTMF
The ACS needs not to support voice band data such as fax or DTMF.
Instead, alternative ways of communication or other RTP payload
format should be considered.
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5.11. Idle noise
The generation of idle channel noise should not be used to indicate
that the call is still active. Instead, in case of transmission
problems an acoustic notification can be given.
5.12. Tandem coding
The ACS needs not to be optimized for tandem coding conditions
because one can assume an end-to-end transmission of IP packets.
Tandem coding might only be used for PSTN gateways and for conference
bridges.
5.13. FEC
RTP support of Forward Error Correction (FEC) needs not to be
considered. Also, support of adding "redundant speech frames", which
have been transmitted in preceding packets, in a RTP packet is not
required. Instead, the redundancy can be added by the encoder which
does this in a more efficient way.
6. Security Considerations
To do.
7. IANA Considerations
To do.
8. References
8.1. Normative References
[ITU-T BS.1534-1] "BS.1534 : Method for the subjective assessment of
intermediate quality levels of coding systems", ITU-T
Recommendation BS.1534-1 (01/03).
[ITU-T G.107] "G.107 : The E-model, a computational model for use in
transmission planning", ITU-T Recommendation G.107 (04/09).
[ITU-T P.800] "P.800 : Methods for subjective determination of
transmission quality", ITU-T Recommendation P.800 (08/96).
[ITU-R BS.1116] "BS.1116 : Methods for the subjective assessment of
small impairments in audio systems including multichannel
sound systems", ITU-R Recommendation BS.1116 (10/97).
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[OMAPoCReq] "Push to talk over Cellular Requirements", Open Mobile
Alliance, Approved Version 1.0, 09 Jun 2006, OMA-RD-PoC-
V1_0-20060609-A.pdf
[TIA-921] TIA-921-A Document Information: "Network Model for
Evaluating Multimedia Transmission Performance Over
Internet Protocol", Publisher: Telecommunications Industry
Association, Publication Date: Jun 18, 2008
[TS26.114-830] 3GPP TS 26.114 V8.3.0, "IP Multimedia Subsystem (IMS);
Multimedia telephony; Media handling and interaction",
Rapporteur: Per Froejdh, Version 8.3.0, 2009-06-12,
RTS/TSGS-0426114v830.
8.2. Informative References
[A2DPV10] Bluetooth SIG, "Advanced Audio Distribution Profile", Audio
Video WG, adopted specification, revision V1.0, May 22th,
2003.
[celt-draft] J-M. Valin, T. Terriberry, G. Maxwell, C. Montgomery,
"Constrained-Energy Lapped Transform (CELT) Codec",
Internet draft, draft-valin-celt-codec-01, work in
progress, July 13, 2009.
[Gurevich2004] Gurevich, M., Chafe, C., Leslie, G., and Tyan, S.,
"Simulation of Networked Ensemble Performance with Varying
Time Delays: Characterization of Ensemble Accuracy",
Proceedings of the 2004 International Computer Music
Conference, Miami, USA, 2004.
[Hoene2005] Hoene, C., and Karl, H., and Wolisz, A., "A perceptual
quality model intended for adaptive VoIP applications",
International Journal of Communication Systems, Wiley,
August 2005.
[SG16 314-WP3] ITU-T SG16, "Agenda and list of documents for Q9/16",
Temporary Document 314-WP3, Received on 2008-04-22 From
Rapporteur Q9/16.
[silk-draft] K. Vos, S. Jensen, K. Soerensen, "SILK Speech Codec",
Internet draft, draft-vos-silk-00.txt, work in progress,
July 6, 2009.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
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9. Acknowledgments
The authors like to thank the various contributors taking part at the
discussion on the Codec BOF mailing list in the period till September
2009. Also, this document is based on the SILK [silk-draft] and CELT
drafts, the internal requirement documents of ITU-T G.718 [SG16 314-
WP3] and the 3GPP document [TS26.114-830].
The author likes to thank Henry Sinnreich for his valuable feedback
and support.
Funding for this draft has been provided by the University of
Tuebingen within the "Projektfoerderung fuer Nachwuchswissen-
schaftler".
This document was prepared using 2-Word-v2.0.template.dot.
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Author's Address
Christian Hoene
University of Tuebingen
WSI-RI
Sand 13
72076 Tuebingen
Germany
Phone: +49 7071 2970532
Email: hoene@ieee.org
Hoene Expires February 17, 2010 [Page 24]
