Internet DRAFT - draft-ietf-payload-rtp-aptx
draft-ietf-payload-rtp-aptx
Internet-Draft J. Lindsay
A/V Transport Payloads Working Group H.Foerster
Intended status: Standards Track APT Ltd
Expires: Aug 05, 2014 February 05, 2014
RTP Payload Format for Standard apt-X and Enhanced apt-X Codecs
draft-ietf-payload-rtp-aptx-05
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.
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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) 2014 IETF Trust and the persons identified as the
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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 . . . . . . . . . . . . . . . . . 8
4.1. Use of Forward Error Correction (FEC) . . . . . . . . . . 8
5. Payload Format . . . . . . . . . . . . . . . . . . . . . . . . 9
5.1. RTP Header Usage . . . . . . . . . . . . . . . . . . . . . 9
5.2. Payload Structure . . . . . . . . . . . . . . . . . . . . 10
5.3. Default Packetization Interval . . . . . . . . . . . . . . 11
5.4. Implementation Considerations . . . . . . . . . . . . . . 11
5.5. Payload Example . . . . . . . . . . . . . . . . . . . . . 11
6. Payload Format Parameters . . . . . . . . . . . . . . . . . . 14
6.1. Media Type Definition . . . . . . . . . . . . . . . . . . 14
6.2. Mapping to SDP . . . . . . . . . . . . . . . . . . . . . . 16
6.2.1. SDP Usage Example . . . . . . . . . . . . . . . . . . 16
6.2.2. Offer/Answer Considerations . . . . . . . . . . . . . 17
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 18
8. Security Considerations . . . . . . . . . . . . . . . . . . . 19
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 20
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 21
10.1. Normative References . . . . . . . . . . . . . . . . . . . 21
10.2. Informative References . . . . . . . . . . . . . . . . . . 21
11. Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 22
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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, which can be licensed from CSR plc. and are 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 optionally
carry synchronisation information and an auxiliary data channel
within the coded audio data without additional overhead. These
mechanisms can, for instance, be used when the IP system is cascaded
with another transportation system and the decoder is acting as a
simple bridge between the two systems. Since auxiliary data channel
and synchronisation information are carried within the coded audio
data without additional overhead, RTP payload format rules do not
change if they are present. Out-of-band signalling is required
however to notify the receiver end when autosync and auxiliary data
have been embedded in the apt-X stream.
Embedded 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 with a single audio channel encoded at Fs =
48kHz, an auxiliary data bit rate of 12 kbit/s can be embedded.
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apt-X further provides a means of stereo pairing apt-X channels so
that the embedded autosync and auxiliary data channel can be shared
across the channel pair. 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 the
Standard or Enhanced apt-X encoder once every 32 uncoded left
channel samples. By default apt-X channels pairing is not enabled.
Out-of-band signalling is required to notify the receiver when the
option is being used.
Standard apt-X and Enhanced apt-X decoders that have not be set up
with the correct embedded autosync, auxiliary data and stereo pairing
information will playout uncoded PCM samples with a loss of decoding
quality. In the case of standard apt-X the loss of quality can be
significant.
Further details on the algorithm operation can be obtained from
CSR plc.
Corporate HQ
Churchill House
Cambridge Business Park
Cowley Road
Cambridge
CB4 0WZ
UK
Tel: +44 1223 692000
Fax: +44 1223 692001
http://www.csr.com
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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
Utilization 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 - As per [RFC3551] Section 4.1
o PT - A dynamic payload type, 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
milliseconds. When an integer number of coded samples per channel
cannot be contained within this 4 milliseconds 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 milliseconds.
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 milliseconds
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 signalling protocol MUST support all
parameters. Implementation can always decide whether they are
capable of communicating based on the entities defined in this
specification.
5.5. 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 4 milliseconds. 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 milliseconds, 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:
Least 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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6. 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 6838 [RFC6838]
6.1. Media Type Definition
Type name: audio
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 that are present in the
audio stream.
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 milliseconds.
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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stereo-channel-pairs:
Defines audio channels that are stereo paired in the stream.
See Section 3. Each pair of audio channels is defined as two
comma-separated values that correspond to channel numbers in
the range 1..channels. Each stereo channel pair is preceded
by a '{', and followed by a '}'. Pairs of audio channels are
separated by a comma. A channel MUST NOT be paired with more
than one other channel. Absence of this parameter signals that
each channel has been independently encoded.
embedded-autosync-channels:
Defines channels that carry embedded autosync. Embedded-
autosync-channels are defined as a list of comma-separated
values that correspond to channel numbers in the range 1..
channels. When a channel is stereo paired, embedded autosync
is shared across channels in the pair. The first channel
as defined in stereo-channel-pairs MUST be specified in the
embedded-autosync-channels list.
embedded-aux-channels:
Defines channels that carry embedded auxiliary data. Embedded-
aux-channels are defined as a list of comma-separated values
that correspond to channel numbers in the range 1..channels.
When a channel is stereo paired, embedded auxiliary data is
shared across channels in the pair. The second channel as
defined in stereo-channel-pairs MUST be specified in the
embedded-autosync-channels list.
Encoding considerations: This media type is framed in RTP and
contains binary data; see Section 4.8 of [RFC6838].
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
Fragment identifier considerations: None
Additional information: none
Person & email address to contact for further information: John
Lindsay email:lindsay@worldcastsystems.com
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Intended usage: Common
Restrictions on usage: This media type depends on RTP framing,
and hence is only defined for transfer via RTP [RFC3550].
Author/Change controller: "IETF Payload Working Group delegated
from the IESG"
6.2. 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
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 parameter "maxptime" when present, MUST be included in the
SDP "a=maxptime" attribute.
The required parameters "variant" and "bitresolution" MUST be
included in the SDP "a=fmtp" attribute.
The optional parameters "stereo-channel-pairs", "embedded-
autosync-channels", "embedded-aux-channels" when present,
MUST be included in the SDP "a=fmtp" attribute.
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 milliseconds of audio represented by one
RTP packet. See Section 6 of [RFC4566].
6.2.1. SDP Usage Examples
Some example SDP session descriptions utilizing apt-X encodings
follow. In these examples, long a=fmtp lines are folded to meet the
column width constraints of this document.
Example 1: A standard apt-X stream that encodes two independent
44.1kHz 16-bit PCM channels into a 4 milliseconds RTP packet.
m=audio 5004 RTP/AVP 98
a=rtpmap:98 aptx/44100/2
a=fmtp:98 variant=standard; bitresolution=16;
a=ptime:4
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Example 2: An enhanced apt-X stream that encodes two 48kHz 24-bit
stereo channels into a 4 milliseconds RTP packet and that carries
both an embedded autosync and auxiliary data channel.
m=audio 5004 RTP/AVP 98
a=rtpmap:98 aptx/48000/2
a=fmtp:98 variant=enhanced; bitresolution=24;
stereo-channel-pairs={1,2}; embedded-autosync-channels=1;
embedded-aux-channels=2
a=ptime:4
Example 3: An enhanced apt-X stream that encodes six 44.1kHz 24-bit
channels into a 6 milliseconds RTP packet. Channels 1,2 and 3,4 are
stereo pairs. Both stereo pairs carry both an embedded autosync and
auxiliary data channel.
m=audio 5004 RTP/AVP 98
a=rtpmap:98 aptx/44100/6
a=fmtp:98 variant=enhanced; bitresolution=24;
stereo-channel-pairs={1,2},{3,4}; embedded-autosync-channels=1,3;
embedded-aux-channels=2,4
a=ptime:6
6.2.2. 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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7. IANA Considerations
One media type (audio/aptx) has been defined and needs registration
in the media types registry. See Section 6.1
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8. 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.
[draft-ietf-avtcore-srtp-vbr-audio] has highlighted potential
security vulnerabilities of Variable Bit Rate (VBR) codecs using
Secure RTP transmission methods. As the Standard apt-X and Enhanced
apt-X codecs are Constant Bit Rate (CBR) codecs, this security
vulnerability is therefore not applicable.
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9. Acknowledgements
This specification was facilitated by earlier documents produced by
Greg Massey, David Trainer, James Hunter and Derrick Rea along with
practical tests carried out by Paul McCambridge of APT Ltd.
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10. References
10.1. 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.
10.2. 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.
[RFC6838] Freed, N.,J. Klensin, and T. Hansen, "Media Type
Specifications and Registration Procedures", BCP 13,
RFC 6838, January 2013.
[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.
[draft-ietf-avtcore-srtp-vbr-audio] Perkins, C., Valins, JM.,
"Guidelines for the Use of Variable Bit Rate Audio with
Secure RTP", draft-ietf-avtcore-srtp-vbr-audio, March 2012.
Lindsay & Foerster Expires Aug 05, 2014 [Page 21]
Internet-Draft apt-X RTP Format Feb 2014
11. Authors' Addresses
John Lindsay
APT Ltd
729 Springfield Road
Belfast
Northern Ireland
BT12 7FP
UK
Phone: +44 2890 677200
Email: Lindsay@worldcastsystems.com
Hartmut Foerster
APT Ltd
729 Springfield Road
Belfast
Northern Ireland
BT12 7FP
UK
Phone: +44 2890 677200
Email: foerster@worldcastsystems.com
Lindsay & Foerster Expires Aug 05, 2014 [Page 22]
