[speex-dev] Updated Speex RTP Internet Draft
philkerr at elec.gla.ac.uk
philkerr at elec.gla.ac.uk
Sun Jun 29 03:12:14 PDT 2003
Hi all,
Please find below an updated Speex Internet Draft document.
It would be good if we could book some time for discussion on Speex at the IETF
meeting in Vienna (scheduled for 14th July). The cutoff for submission is
9:00am EDT, (GMT -04:00), 30th June.
Comments and feedback welcomed!
Regards
Phil
-----------------8<--------------------------------------------8<-------------------------------
<p><p><p>Internet Engineering Task Force Greg Herlein
Internet Draft Jean-Marc Valin
draft-herlein-speex-rtp-profile-01 Simon Morlat
June 30, 2003 Roger Hardiman
Expires: January 30, 2004 Phil Kerr
<p> RTP Payload Format for the Speex Codec
Status of this Memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
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.
Internet-Drafts are draft documents valid for a maximum of six
months and may be updated, replaced, or obsoleted by other
documents at any time. It is inappropriate to use Internet-Drafts
as reference material or to cite them other than as "work in
progress".
The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt
To view the list Internet-Draft Shadow Directories, see
http://www.ietf.org/shadow.html.
<p>Copyright Notice
Copyright (C) The Internet Society (2003). All Rights Reserved.
<p>Abstract
Speex is an open-source, patent-free voice codec suitable for use in
Voice over IP (VoIP) type applications. This document describes the
payload format for Speex generated bit streams within an RTP packet.
Also included here are the necessary details for the use of Speex with
the Session Description Protocol (SDP) and a preliminary method
of using Speex within H.323 applications.
<p>1. Conventions used in this document
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 [5].
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<p>2. Overview of the Speex Codec
Speex is based on the CELP [12] encoding technique with support for
either narrowband (nominal 8kHz), wideband (nominal 16kHz) or
ultra-wideband (nominal 32kHz), and (non-optimal) rates up to 48 kHz
sampling also available. The main characteristics can be summerized
as follows:
o Free software/open-source, royalty-free
o Integration of wideband and narrowband in the same bit-stream
o Wide range of bit-rates available
o Dynamic bit-rate switching and variable bit-rate (VBR)
o Voice Activity Detection (VAD, integrated with VBR)
o Variable complexity
<p>3. RTP payload format for Speex
For RTP based transportation of Speex encoded audio the standard
RTP header [2] is followed by a four bit header block which is
used to indicate the presence of one or more payload data blocks.
An optional padding terminator may also be used.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTP Header |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| one or more frames of Speex .... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| one or more frames of Speex .... | padding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
<p>3.1 RTP Header
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC |M| PT | sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| contributing source (CSRC) identifiers |
| ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
<p> The RTP header begins with an octet of fields (V, P, X, and CC) to
support specialized RTP uses (see [8] and [9] for details). For
Speex the following values are used.
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<p> Version (V): 2 bits
This field identifies the version of RTP. The version
used by this specification is two (2).
Padding (P): 1 bit
If the padding bit is set, the packet contains one or more
additional padding octets at the end which are not part of
the payload. P is set if the total packet size is less than
the MTU.
Extension (X): 1 bit
If the extension, X, bit is set, the fixed header MUST be
followed by exactly one header extension, with a format defined
in Section 5.3.1. of [8],
CSRC count (CC): 4 bits
The CSRC count contains the number of CSRC identifiers.
Marker (M): 1 bit
The M bit indicates if the packet contains comfort noise. This
field is used in conjunction with the cng SDP attribute and is
detailed further in section 5 below. In normal usage this bit
is set if the packet contains comfort noise.
Payload Type (PT): 7 bits
An RTP profile for a class of applications is expected to assign
a payload type for this format, or a dynamically allocated
payload type SHOULD be chosen which designates the payload as
Speex.
Sequence number: 16 bits
The sequence number increments by one for each RTP data packet
sent, and may be used by the receiver to detect packet loss and
to restore packet sequence. This field is detailed further in
[2].
Timestamp: 32 bits
A timestamp representing the sampling time of the first sample of
the first Speex packet in the RTP packet. The clock frequency
MUST be set to the sample rate of the encoded audio data.
Speex uses 20 msec frames and a variable sampling rate clock.
The RTP timestamp MUST be in units of 1/X of a second where X
is the sample rate used. Speex uses a nominal 8kHz sampling rate
for narrowband use, a nominal 16kHz sampling rate for wideband use,
and a nominal 32kHz sampling rate for ultra-wideband use.
SSRC/CSRC identifiers:
These two fields, 32 bits each with one SSRC field and a maximum
of 16 CSRC fields, are as defined in [2].
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<p>3.2 Speex payload
<p> For the purposes of packetizing the bit stream in RTP, it is only
necessary to consider the sequence of bits as output by the Speex
encoder [11], and present the same sequence to the decoder. The
payload format described here maintains this sequence.
A typical Speex frame, encoded at the maximum bitrate, is approx.
110 octets and the total number of Speex frames SHOULD be kept
less than the path MTU to prevent fragmentation. Speex frames MUST
NOT be fragmented across multiple RTP packets,
An RTP packet MAY contain Speex frames of the same bit rate or of
varying bit rates, since the bit-rate for a frame is conveyed in
band with the signal.
The encoding and decoding algorithm can change the bit rate at any
20ms frame boundary but the bit rate change notification is
provided in-band with the bit stream. Each frame contains both
"mode" (narrowband, wideband or ultra-wideband) and "sub-mode"
(bit-rate) information in the bit stream. No out-of-band
notification is required for the decoder to process changes in the
bit rate sent by the encoder.
It is RECOMMENDED that values of 8000 or 16000 be used for normal
internet telephony applications, though the sample rate is
supported at rates as low as 6000 Hz and as high as 48 kHz.
The RTP payload MUST be padded to provide an integer number of
octets as the payload length. These padding bits are set in
the pattern of a zero, LSB aligned network byte order, followed
by all ones (until the end of the octet). This padding is only
required for the last frame in the packet, and only to ensure
the packet contents ends on an octet boundary.
<p>3.2.1 Example Speex packet
In the example below we have a single Speex frame with 5 bits
of padding to ensure the packet size falls on an octet boundary.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC |M| PT | sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| contributing source (CSRC) identifiers |
| ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ..speex data.. |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ..speex data.. |0 1 1 1 1|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
<p>3.4 Multiple Speex frames in a RTP packet
Below is an example of two Speex frames contained within one RTP
packet. The Speex frame length in this example fall on an octet
boundary so there is no padding.
Speex codecs [11] are able to detect the the bitrate from the
payload and are responsible for detecting the 20 msec boundaries
between each frame.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC |M| PT | sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| contributing source (CSRC) identifiers |
| ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ..speex data.. |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ..speex data.. | ..speex data.. |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ..speex data.. |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
<p>4. MIME registration of Speex
Full definition of the MIME type for Speex will be part of the Ogg
Vorbis MIME type definition application [10].
MIME media type name: audio
MIME subtype: speex
Required parameters: to be included in the Ogg MIME specification.
Optional parameters:
Encoding considerations:
Security Considerations:
See Section 6 of RFC 3047.
Interoperability considerations: none
Published specification:
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<p> Applications which use this media type:
Additional information: none
Person & email address to contact for further information:
Greg Herlein <gherlein at herlein.com>
Jean-Marc Valin <jean-marc.valin at hermes.usherb.ca>
Intended usage: COMMON
Author/Change controller:
Author: Greg Herlein <gherlein at herlein.com>
Change controller: Greg Herlein <gherlein at herlein.com>
This transport type signifies that the content is to be interpreted
according to this document if the contents are transmitted over RTP.
Should this transport type appear over a lossless streaming protocol
such as TCP, the content encapsulation should be interpreted as an
Ogg Stream in accordance with RFC 3534, with the exception that the
content of the Ogg Stream may be assumed to be Speex audio and
Speex audio only."
<p>5. SDP usage of Speex
When conveying information by SDP [4], the encoding name MUST be
set to "speex". An example of the media representation in SDP for
offering a single channel of Speex at 8000 samples per second might
be:
m=audio 8088 RTP/AVP 97
a=rtpmap:97 speex/8000
Note that the RTP payload type code of 97 is defined in this media
definition to be 'mapped' to the speex codec at an 8kHz sampling
frequency using the 'a=rtpmap' line. Any number from 96 to 127
could have been chosen (the allowed range for dynamic types). The
value of the sampling frequency is typically 8000 for narrow band
operation, 16000 for wide band operation, and 32000 for ultra-wide
band operation.
If for some reason the offerer has bandwidth limitations, the client
may use the "b=" header, as explained in SDP [4]. The following example
illustrates the case where the offerer cannot receive more than
10 kbit/s.
m=audio 8088 RTP/AVP 97
b=AS:10
a=rtmap:97 speex/8000
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<p> In this case, if the remote part agrees, it should configure its
Speex encoder so that it does not use modes that produce more than
10 kbit/s. Note that the "b=" constraint also applies on all
payload types that may be proposed in the media line ("m=").
An other way to make recommendations to the remote Speex encoder
is to use its specific parameters via the a=fmtp: directive. The
following parameters are defined for use in this way:
ptime: duration of each packet in milliseconds.
sr: actual sample rate in Hz.
ebw: encoding bandwidth - either 'narrow' or 'wide' or
'ultra' (corresponds to nominal 8000, 16000, and
32000 Hz sampling rates).
vbr: variable bit rate - either 'on' 'off' or 'vad'
(defaults to off). If on, variable bit rate is
enabled. If off, disabled. If set to 'vad' then
constant bit rate is used but silence will be encoded
with special short frames to indicate a lack of voice
for that period.
cng: comfort noise generation - either 'on' or 'off'. If
off then silence frames will be silent; if 'on' then
those frames will be filled with comfort noise.
mode: Speex encoding mode. Can be {1,2,3,4,5,6,any}
defaults to 3 in narrowband, 6 in wide and ultra-wide.
penh: use of perceptual enhancement. 1 indicates
to the decoder that perceptual enhancement is recommended,
0 indicates that it is not. Defaults to on (1).
Examples:
m=audio 8008 RTP/AVP 97
a=rtpmap:97 speex/8000
a=fmtp:97 mode=4
This examples illustrate an offerer that wishes to receive
a Speex stream at 8000Hz, but only using speex mode 3.
The offerer may suggest to the remote decoder to activate
its perceptual enhancement filter like this:
m=audio 8088 RTP/AVP 97
a=rtmap:97 speex/8000
a=fmtp:97 penh=1
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<p> Several Speex specific parameters can be given in a single
a=fmtp line provided that they are separated by a semi-colon:
a=fmtp:97 mode=any;penh=1
The offerer may indicate that it wishes to send variable bit rate
frames with comfort noise:
m=audio 8088 RTP/AVP 97
a=rtmap:97 speex/8000
a=fmtp:97 vbr=on;cng=on
The "ptime" attribute is used to denote the packetization
interval (ie, how many milliseconds of audio is encoded in a
single RTP packet). Since Speex uses 20 msec frames, ptime values
of multiples of 20 denote multiple Speex frames per packet.
Values of ptime which are not multiples of 20 MUST be ignored
and clients MUST use the default value of 20 instead.
In the example below the ptime value is set to 40, indicating that
there are 2 frames in each packet.
m=audio 8008 RTP/AVP 97
a=rtpmap:97 speex/8000
a=ptime:40
Note that the ptime parameter applies to all payloads listed
in the media line and is not used as part of an a=fmtp directive.
Values of ptime not multiple of 20 msec are meaningless, so the
receiver of such ptime values MUST ignore them. If during the
life of an RTP session the ptime value changes, when there are
multiple Speex frames for example, the SDP value must also reflect
the new value.
Care must be taken when setting the value of ptime so that the
RTP packet size does not exceed the path MTU.
<p>6. ITU H.323/H.245 Use of Speex
Application is underway to make Speex a standard ITU codec.
However, until that is finalized, Speex MAY be used in H.323 [6] by
using a non-standard codec block definition in the H.245 [7] codec
capability negotiations.
<p>6.1 NonStandardMessage format
For Speex use in H.245 [7] based systems, the fields in the
NonStandardMessage should be:
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<p> t35CountryCode = Hex: B5
t35Extension = Hex: 00
manufacturerCode = Hex: 0026
[Length of the Binary Sequence (8 bit number)]
[Binary Sequence consisting of an ASCII string, no NULL terminator]
The binary sequence is an ascii string merely for ease of use.
The string is not null terminated. The format of this string is
speex [optional variables]
The optional variables are identical to those used for the SDP
a=fmtp strings discussed in section 5 above. The string is built
to be all on one line, each key-value pair separated by a
semi-colon. The optional variables MAY be omitted, which causes
the default values to be assumed. They are:
ebw=narrow;mode=3;vbr=off;cng=off;ptime=20;sr=8000;penh=no;
The fifth byte of the block is the length of the binary sequence.
NOTE: this method can result in the advertising of a large number
of Speex 'codecs' based on the number of variables possible. For
most VoIP applications, use of the default binary sequence of
'speex' is RECOMMENDED to be used in addition to all other options.
This maximizes the chances that two H.323 based applications that
support Speex can find a mutual codec.
<p>6.2 RTP Payload Types
Dynamic payload type codes MUST be negotiated 'out-of-band'
for the assignment of a dynamic payload type from the
range of 96-127. H.323 applications MUST use the H.245
H2250LogicalChannelParameters encoding to accomplish this.
<p>7. Security Considerations
RTP packets using the payload format defined in this specification
are subject to the security considerations discussed in the RTP
specification [2], and any appropriate RTP profile. This implies
that confidentiality of the media streams is achieved by encryption.
Because the data compression used with this payload format is applied
end-to-end, encryption may be performed after compression so there is
no conflict between the two operations.
A potential denial-of-service threat exists for data encodings using
compression 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
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<p> be overloaded. However, this encoding does 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.
<p>8. Normative References
1. Bradner, S., "The Internet Standards Process -- Revision 3", BCP
9, RFC 2026, October 1996.
2. Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP:
A Transport Protocol for real-time applications", RFC 1889,
January 1996.
3. Freed, N. and N. Borenstein, "Multipurpose Internet Mail
Extensions (MIME) Part One: Format of Internet Message Bodies",
RFC 2045, November 1996.
4. Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
5. Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
6. ITU-T Recommendation H.323. "Packet-based Multimedia
Communications Systems," 1998.
7. ITU-T Recommendation H.245 (1998), "Control of communications
between Visual Telephone Systems and Terminal Equipment".
8. RTP: A transport protocol for real-time applications. Work
in progress, draft-ietf-avt-rtp-new-11.txt.
9. RTP Profile for Audio and Video Conferences with Minimal
Control. Work in progress, draft-ietf-avt-profile-new-12.txt.
10. L. Walleij, "The application/ogg Media Type", RFC 3534, May
2003.
<p>8.1 Informative References
11. Speexenc/speexdec, reference command-line encoder/decoder,
Speex website, http://www.speex.org/
12. CELP, U.S. Federal Standard 1016. National Technical
Information Service (NTIS) website, http://www.ntis.gov/
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<p>9. Acknowledgments
The authors would like to thank Equivalence Pty Ltd of Australia
for their assistance in attempting to standardize the use of Speex
in H.323 applications, and for implementing Speex in their open
source OpenH323 stack. The authors would also like to thank Brian
C. Wiles <brian at streamcomm.com> of StreamComm for his assistance in
developing the proposed standard for Speex use in H.323
applications.
The authors would also like to thank the following members of the
Speex and AVT communities for their input: Ross Finlayson,
Federico Montesino Pouzols, Henning Schulzrinne, Magnus Westerlund.
<p>10. Author's Address
Greg Herlein <gherlein at herlein.com>
2034 Filbert Street
San Francisco, CA
United States 94123
Jean-Marc Valin <jean-marc.valin at hermes.usherb.ca>
Department of Electrical and Computer Engineering
University of Sherbrooke
2500 blvd Universit‰
Sherbrooke, Quebec, Canada, J1K 2R1
<p> Simon MORLAT <simon.morlat at linphone.org>
35, av de Vizille App 42
38000 GRENOBLE
FRANCE
<p> Roger Hardiman <roger at freebsd.org>
49 Nettleton Road
Cheltenham
Gloucestershire
GL51 6NR
England
<p> Phil Kerr <philkerr at elec.gla.ac.uk>
Centre for Music Technology
University of Glasgow
Glasgow
G12 8LT
Scotland
<p>10. Full Copyright Statement
Copyright (C) The Internet Society (2003). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
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followed, or as required to translate it into languages other than
English.
The limited permissions granted above are perpetual and will not be
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This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
<p>Acknowledgement
Funding for the RFC Editor function is currently provided by the
Internet Society.
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