RTP Payload Format for Standard apt-X and Enhanced apt-X Codecs
draft-rea-payload-rtp-aptx-01
The information below is for an old version of the document.
| Document | Type |
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|---|---|---|---|
| Authors | James Hunter , Derrick Rea | ||
| Last updated | 2011-10-07 (Latest revision 2011-04-18) | ||
| RFC stream | (None) | ||
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draft-rea-payload-rtp-aptx-01
Internet-Draft J. Hunter
A/V Transport Payloads Working Group D. Rea
Intended status: Standards Track APT Ltd
Expires: April 7, 2012 October 7, 2011
RTP Payload Format for Standard apt-X and Enhanced apt-X Codecs
draft-rea-payload-rtp-aptx-01
Abstract
This document specifies a scheme for packetizing Standard apt-X, or
Enhanced apt-X, encoded audio data into Real-time Transport Protocol
(RTP) packets. The document describes a payload format that permits
transmission of multiple related audio channels in a single RTP
payload, and a means of establishing Standard apt-X and Enhanced
apt-X connections through the Session Description Protocol (SDP).
Status of this Memo
This Internet-Draft is submitted to IETF in full conformance with the
provisions of BCP 78 and BCP 79.
Comments are solicited and should be addressed to the A/V Transport
Payloads working group's mailing list at payload@ietf.org and/or the
authors.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet-
Drafts.
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http://www.ietf.org/shadow.html.
This Internet-Draft will expire on April 7, 2012.
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Submission Compliance for Internet-Drafts.
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Copyright and License 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 . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 5
3. Standard apt-X and Enhanced apt-X Codecs . . . . . . . . . . . 6
4. Payload Format Capabilities . . . . . . . . . . . . . . . . . 7
4.1. Use of Forward Error Correction (FEC) . . . . . . . . . . 7
5. Payload Format . . . . . . . . . . . . . . . . . . . . . . . . 8
5.1. RTP Header Usage . . . . . . . . . . . . . . . . . . . . . 8
5.2. Payload Structure . . . . . . . . . . . . . . . . . . . . 9
5.3. Default Packetization Interval . . . . . . . . . . . . . . 10
5.4. Implementation Considerations . . . . . . . . . . . . . . 10
6. Payload Example . . . . . . . . . . . . . . . . . . . . . . . 10
7. Payload Format Parameters . . . . . . . . . . . . . . . . . . 13
7.1. Media Type Definition . . . . . . . . . . . . . . . . . . 13
7.2. Mapping to SDP . . . . . . . . . . . . . . . . . . . . . . 14
7.2.1. SDP Usage Example . . . . . . . . . . . . . . . . . . 15
7.2.2. Offer/Answer Considerations . . . . . . . . . . . . . 16
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 17
9. Security Considerations . . . . . . . . . . . . . . . . . . . 18
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 19
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
11.1. Normative References . . . . . . . . . . . . . . . . . . . 20
11.2. Informative References . . . . . . . . . . . . . . . . . . 20
12. Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 21
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1. Introduction
This document specifies the payload format for packetization of audio
data, encoded with the Standard apt-X or Enhanced apt-X audio coding
algorithms, into the Real-time Transport Protocol (RTP). [RFC3550].
The document outlines some conventions, a brief description of the
operating principles of the audio codecs, and the payload format
capabilities. The RTP payload format is detailed and a relevant
example of the format is provided. The media type, its mappings to
SDP [RFC4566] and its usage in the SDP offer/answer model are also
specified. Finally, some security considerations are outlined.
This document registers a media type (audio/aptx) for the RTP payload
format for the Standard apt-X and Enhanced apt-X audio codecs.
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2. Conventions
This document uses the normal IETF bit-order representation. Bit
fields in figures are read left to right and then down. The leftmost
bit in each field is the most significant. The numbering starts from
0 and ascends, where bit 0 will be the most significant.
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 [RFC2119].
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3. Standard apt-X and Enhanced apt-X Codecs
Standard apt-X and Enhanced apt-X are proprietary audio coding
algorithms, licensed by CSR plc and widely deployed in a variety of
audio processing equipment. For commercial reasons, the detailed
internal operations of these algorithms are not described in
standards or reference documents. However, the data interfaces to
implementations of these algorithms are very simple, and allow easy
RTP packetization of data coded with the algorithms, without a
detailed knowledge of the actual coded audio stream syntax.
Both the Standard apt-X and Enhanced apt-X coding algorithms are
based on Adaptive Differential Pulse Code Modulation principles.
They produce a constant coded bit rate that is scaled according to
the sample frequency of the uncoded audio. This constant rate is 1/4
of the bit rate of the uncoded audio, irrespective of the resolution
(number of bits) used to represent an uncoded audio sample. For
example, a 1.536 Mbit/s stereo audio stream, composed of 2 channels
of 16-bit Pulse Code Modulated (PCM) audio that is sampled at a
frequency of 48 kHz, is encoded at 384 kbit/s.
Standard apt-X and Enhanced apt-X do not enforce a coded frame
structure, and the coded data forms a continuous coded sample stream,
with each coded sample capable of regenerating 4 PCM samples when
decoded. The Standard apt-X algorithm encodes 4 successive 16-bit
PCM samples from each audio channel into a single 16-bit coded sample
per audio channel. The Enhanced apt-X algorithm encodes 4 successive
16-bit or 24-bit PCM samples from each audio channel and respectively
produces a single 16-bit or 24-bit coded sample per channel. The
same RTP packetisation rules apply for each of these algorithmic
variations.
Standard apt-X and Enhanced apt-X coded data streams can carry an
auxiliary data channel and synchronisation information within the
coded audio data without additional overhead. As a result, when
carrying auxiliary data or synchronisation information, RTP payload
format rules do not change.
Auxiliary data is typically used to transport non-audio data, and
timecode information for synchronisation with video. The bit rate
of the auxiliary data channel is 1/4 of the sample frequency. For
example at Fs = 48 kHz, the bit rate of the auxiliary data stream
is 12 kbit/s. In the case of a 1.536 Mbit/s stereo audio stream,
composed of 2 channels of 16-bit PCM audio that is sampled at
48 kHz, a byte of auxiliary data would typically be fed into a
Standard or Enhanced apt-X encoder once every 24 left channel
samples.
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4. Payload Format Capabilities
This RTP payload format carries an integer number of Standard apt-X
or Enhanced apt-X coded audio samples. When multiple related audio
channels are being conveyed within the payload, each channel
contributes the same integer number of apt-X coded audio samples to
the total carried by the payload.
4.1. Use of Forward Error Correction (FEC)
Standard apt-X and Enhanced apt-X do not inherently provide any
mechanism for adding redundancy or error-control coding into the
coded audio stream. Generic forward error correction schemes for RTP
such as RFC 2198 [RFC2198] and RFC 5109 [RFC5109] can be used to add
redundant information to Standard apt-X and Enhanced apt-X RTP packet
streams, making them more resilient to packet losses at the expense
of a higher bit rate.
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5. Payload Format
The Standard apt-X and Enhanced apt-X algorithms encode 4 successive
PCM samples from each audio channel and produce a single compressed
sample for each audio channel. The encoder MUST be presented with
an integer number S of input audio samples, where S is an arbitrary
multiple of 4. The encoder will produce exactly S/4 coded audio
samples. Since each coded audio sample is either 16 or 24 bits, the
amount of coded audio data produced upon each invocation of the
encoding process will be an integer number of bytes. RTP
packetization of the encoded data SHALL be on a byte-by-byte basis.
5.1. RTP Header Usage
Utilisation of the Standard apt-X and Enhanced apt-X coding
algorithms does not create any special requirements with respect to
the contents of the RTP packet header. Other RTP packet header
fields are defined as follows.
o V - As per [RFC3550]
o P - As per [RFC3550]
o X - As per [RFC3550]
o CC - As per [RFC3550]
o M - This payload format defines no use for this bit. Senders
SHOULD set this bit to zero in each outgoing packet.
o PT - A dynamic payload type, i.e. one within the range [96..127],
MUST be used. [RFC3551]
o SN - As per [RFC3550]
o Timestamp - As per [RFC3550]. The RTP timestamp reflects the
instant at which the first audio sample in the packet was sampled,
that is, the oldest information in the packet.
Header field abbreviations are defined as follows.
V - Version Number
P - Padding
X - Extensions
CC - Count of contributing sources
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M - Marker
PT - Payload Type
PS - Payload Structure
5.2. Payload Structure
The RTP payload data for Standard apt-X and Enhanced apt-X MUST be
structured as follows.
Standard and Enhanced apt-X coded samples are packed contiguously
into payload octets in "network byte order", also known as big-endian
order and starting with the most significant bit. Coded samples are
packed into the packet in time sequence beginning with the oldest
coded sample. An integer number of coded samples MUST be within the
same packet.
When multiple channels of Standard and E-APTX coded audio, such as
in a stereo program, are multiplexed into a single RTP stream, the
coded samples from each channel, at a single sampling instant, are
interleaved into a coded sample block according to the following
standard audio channel ordering, [RFC3551]. Coded sample blocks are
then packed into the packet in time sequence beginning with the
oldest coded sample block.
l left
r right
c center
S surround
F front
R rear
channels description channel
1 2 3 4 5 6
_________________________________________________
2 stereo l r
3 l r c
4 l c r S
5 Fl Fr Fc Sl Sr
6 l lc c r rc S
For the two-channel encoding example, the sample sequence is (left
channel, first sample), (right channel, first sample), (left channel,
second sample), (right channel, second sample). Coded Samples for all
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channels belonging to a single coded sampling instant MUST be
contained in the same packet. All channels in the same RTP stream
MUST be sampled at the same frequency.
5.3. Default Packetization Interval
The default packetization interval MUST have a duration of 4 ms.
When an integer number of coded samples per channel can not be
contained within this 4ms interval, the default packet interval MUST
be rounded down to the nearest packet interval that can contain a
complete integer set of coded samples. For example when encoding
audio with either Standard or Enhanced apt-X, sampled at 11025 Hz,
22050 Hz, or 44100 Hz, the packetization interval MUST be rounded
down to 3.99 ms.
The packetization interval sets limits on the end-to-end delay;
shorter packets minimize the audio delay through a system at the
expense of increased bandwidth while longer packets introduce
less header overhead but increase delay and make packet loss
more noticeable. A default packet interval of 4 ms maintains an
acceptable ratio of payload to header bytes and minimizes
the end-to-end delay to allow viable interactive apt-X based
applications. All implementations MUST support this default
packetization interval.
5.4. Implementation Considerations
An application implementing this payload format MUST understand all
the payload parameters that are defined in this specification. Any
mapping of these parameters to a signaling protocol MUST support all
parameters. Implementation can always decide whether they are
capable of communicating based on the entities defined in this
specification.
6. Payload Example
As an example payload format, consider the transmission of an
arbitrary 5.1 audio signal consisting of 6 channels of 24-bit PCM
data, sampled at a rate of 48 kHz and packetized on a RTP packet
interval of 4ms. The total bit rate before audio coding is
6 * 24 * 48000 = 6.912 Mbits/s. Applying Enhanced apt-X coding,
with a coded sample size of 24 bits, results in a transmitted coded
bit rate of 1/4 of the uncoded bit rate, i.e. 1.728 Mbit/s. On packet
intervals of 4 ms, packets contain 864 bytes of encoded data that
contain 48 Enhanced apt-X coded samples per channel.
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For the example format, the diagram below shows how coded samples
from each channel are packed into a sample block and how sample
blocks 1, 2, and 48 are subsequently packed into the RTP packet.
C:
Channel index: Left (l) = 1, left centre (lc) = 2, centre
(c) = 3, right (r) = 4, right centre (rc) = 5, surround (S) = 6.
T:
Sample Block time index: The first sample block is 1, the final
sample is 48.
S(C)(T):
The Tth sample from channel C
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(1)(1) | S(2)(1) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(2)(1) | S(3)(1) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(3)(1) | S(4)(1) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(5)(1) | S(6)(1) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(6)(1) | S(1)(2) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(2)(2) | S(3)(2) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(4)(2) | S(5)(2) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(5)(2) | S(6)(2) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(6)(2) | S(1)(3) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(6)(47) | S(1)(48) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(1)(48) | S(2)(48) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(3)(48) | S(4)(48) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(4)(48) | S(5)(48) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| S(5)(48) | S(6)(48) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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For the example format, the diagram below indicates the order that
coded bytes are packed into the packet payload in terms of sample
byte significance. The following abbreviations are used.
MSB:
Most Significant Byte of a 24-bit coded sample
MB:
Middle Byte of a 24-bit coded sample
LSB:
Most Significant Byte of a 24-bit coded sample
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| MSB | MB | LSB | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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7. Payload Format Parameters
This RTP payload format is identified using the media type audio/
aptx, which is registered in accordance with RFC 4855 [RFC4855] and
using the template of RFC 4288 [RFC4288]
7.1. Media Type Definition
Registration of media subtype audio/aptx.
MIME media type name: audio
MIME subtype name: aptx
Required parameters:
rate:
RTP timestamp clock rate, which is equal to the sampling rate
in Hz. -- RECOMMENDED values for rate are 8000, 11025, 16000,
22050, 24000, 32000, 44100 and 48000 samples per second. Other
values are permissible.
channels:
The number of logical audio channels are present in an audio
stream. Defaults to 2 (stereo).
variant:
The variant of apt-X (i.e. Standard or Enhanced) that is being
used. The following variants can be signalled:
variant=standard
variant=enhanced
bitresolution:
The number of bits used by the algorithm to encode 4 PCM
samples. This value MAY only be set to 16 for Standard apt-X
and 16 or 24 for Enhanced apt-X.
Optional parameters:
ptime:
The recommended length of time (in milliseconds) represented by
the media in a packet. Defaults to 4 ms. See Section 6 of
[RFC4566].
maxptime:
The maximum length of time (in milliseconds) that can be
encapsulated in a packet. See Section 6 of [RFC4566].
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aux:
Indicates the transportation method of the auxiliary data
discussed in Section 3. Allowed values are:
aux=off
No aux data present.
aux=embedded
Aux data embedded within the primary audio stream.
aux=out-of-bound-aux
Aux data transmitted on a separate ip stream. If this value
is set then aux-port parameter MUST also be present.
If the aux parameter is omitted then it is assumed that no
auxiliary data is available.
aux-port:
Indicates the port on which aux data can be received. This
parameter MUST be present when aux value is set to
"out-of-bound-aux", in all other cases this value MUST NOT be
used.
Encoding considerations: This type is only defined for transfer
via RTP [RFC3550].
Security considerations: See Section 5 of [RFC4855] and Section 4
of [RFC4856].
Interoperability considerations: none
Published specification: RFC XXXX
Applications which use this media type: Audio streaming
Additional information: none
Person & email address to contact for further information: Derrick
Rea email:rea@worldcastsystems.com
Intended usage: COMMON
Author/Change controller: Derrick Rea
7.2. Mapping to SDP
The information carried in the media type specification has a
specific mapping to fields in the Session Description Protocol (SDP)
[RFC4566] that is commonly used to describe RTP sessions. When SDP
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is used to describe sessions the media type mappings are as follows.
The type name ("audio") goes in SDP "m=" as the media name.
The subtype name ("aptx") goes in SDP "a=rtpmap" as the encoding
name.
The parameter "rate" also goes in "a=rtpmap" as clock rate.
The parameter "channels" also goes in "a=rtpmap" as channel count.
The required parameters "variant" and "bitresolution" MUST be
included in the SDP "a=fmtp" attribute and MUST follow the
delivery-method that applies.
The optional parameters "aux", "aux-port", and "maxptime" when
present, MUST be included in the SDP "a=fmtp" attribute and MUST
follow the delivery-method that applies.
The parameter "ptime", when present, goes in a separate SDP
attribute field and is signalled as "a=ptime:<value>", where
<value> is the number of millseconds of audio represented by one
RTP packet. See Section 6 of [RFC4566].
7.2.1. SDP Usage Example
The following example shows a basic SDP description of a single audio
stream. The first configuration packet is inlined in the SDP, other
configurations could be fetched at any time from the first provided
uri, or all the known configuration could be downloaded using the
second uri.
c=IN IP4 192.0.2.24
m=audio 5004 RTP/AVP 98
a=rtpmap:98 aptx/44100/2
a=fmtp:98 variant=enhanced; bitresolution=24;
aux=out-of-bound-aux; aux-port=4000;
a=ptime:4
Note that parameter names are case-insensitive both in media types
and in the mapping to the SDP a=fmtp attribute.
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7.2.2. Offer/Answer Considerations
The only negotiable parameter is the delivery method. All other
parameters are declarative. The offer, as described in [RFC3264],
may contain a large number of delivery methods per single fmtp
attribute, the answerer MUST remove every delivery method and
configuration uri not supported. Apart from this exceptional case,
all parameters MUST NOT be altered on answer.
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8. IANA Considerations
One media type (audio/aptx) has been defined and needs registration
in the media types registry. See Section 7.1
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9. Security Considerations
RTP packets using the payload format defined in this specification
are subject to the security considerations discussed in the RTP
specification [RFC3550], and any appropriate RTP profile (for example
[RFC3551]). This implies that confidentiality of the media streams
is achieved by encryption. Because the audio coding used with this
payload format is applied end-to-end, encryption may be performed
after audio coding so there is no conflict between the two
operations. A potential denial-of-service threat exists for audio
coding techniques that have non-uniform receiver-end computational
load. The attacker can inject pathological datagrams into the stream
which are complex to decode and cause the receiver to be overloaded.
However, the Standard apt-X and Enhanced apt-X audio coding
algorithms do not exhibit any significant non-uniformity. As with
any IP-based protocol, in some circumstances a receiver may be
overloaded simply by the receipt of too many packets, either desired
or undesired. Network-layer authentication may be used to discard
packets from undesired sources, but the processing cost of the
authentication itself may be too high. In a multicast environment,
pruning of specific sources may be implemented in future versions of
IGMP [RFC3376] and in multicast routing protocols to allow a receiver
to select which sources are allowed to reach it.
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10. Acknowledgements
This specification was facilitated by earlier documents produced
by Greg Massey and David Trainer, and practical tests carried out
by Paul McCambridge of APT Ltd.
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11. References
11.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC3551] H. Schulzrinne, "RTP profile for audio and video
conferences with minimal control", RFC 3551, July 2003.
11.2. Informative References
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997.
[RFC3376] Cain, B., Deering, S., Kouvelas, I., Fenner, B., and A.
Thyagarajan, "Internet Group Management Protocol, Version
3", RFC 3376, October 2002.
[RFC4288] Freed, N. and J. Klensin, "Media Type Specifications and
Registration Procedures", BCP 13, RFC 4288, December 2005.
[RFC4855] Casner, S., "Media Type Registration of RTP Payload
Formats", RFC 4855, February 2007.
[RFC4856] Casner, S., "Media Type Registration of Payload Formats in
the RTP Profile for Audio and Video Conferences",
RFC 4856, February 2007.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, December 2007.
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12. Authors' Addresses
James Hunter
APT Ltd
729 Springfield Road
Belfast
Northern Ireland
BT12 7FP
UK
Phone: +44 2890 677200
Email: Hunter@worldcastsystems.com
Derrick Rea
APT Ltd
729 Springfield Road
Belfast
Northern Ireland
BT12 7FP
UK
Phone: +44 2890 677200
Email: Rea@worldcastsystems.com
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