AVT P. Luthi
Internet-Draft Tandberg
Expires: December 22, 2005 R. Even, Ed.
Polycom
June 20, 2005
RTP Payload Format for ITU-T Recommendation G.722.1
draft-even-avt-rfc3047-bis-01.txt
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Copyright (C) The Internet Society (2005).
Abstract
International Telecommunication Union (ITU-T) Recommendation G.722.1
is a wide-band audio codec. This document describes the payload
format for including G.722.1 generated bit streams within an RTP
packet. The document also describe the syntax and semantics of the
SDP parameters needed to support G.722.1 audio codec.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. RTP payload format for G.722.1 . . . . . . . . . . . . . . . . 5
3.1 Multiple G.722.1 frames in a RTP packet . . . . . . . . . 7
3.2 Computing the number of G.722.1 frames . . . . . . . . . . 7
4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
4.1 Registration of MIME media type audio/G7221 . . . . . . . 8
5. SDP Parameters . . . . . . . . . . . . . . . . . . . . . . . . 10
5.1 Usage with the SDP Offer Answer Model . . . . . . . . . . 10
6. Security Considerations . . . . . . . . . . . . . . . . . . . 11
7. Changes from RFC 3047 . . . . . . . . . . . . . . . . . . . . 12
8. Normative References . . . . . . . . . . . . . . . . . . . . . 12
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 12
Intellectual Property and Copyright Statements . . . . . . . . 14
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1. Introduction
ITU-T G.722.1[ITU.G7221] is a low complexity coder, it compresses
50Hz - 14kHz audio signals into one of the following bit rates, 24
kbit/s, 32 kbit/s or 48 kbit/s.
The coder may be used for speech, music and other types of audio.
Some of the applications for which this coder is suitable are:
o Real-time communications such as videoconferencing and telephony.
o Streaming audio
o Archival and messaging
A fixed frame size of 20 ms is used, and for any given bit rate the
number of bits in a frame is a constant.
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2. Terminology
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 RFC2119[RFC2119] and
indicate requirement levels for compliant RTP implementations.
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3. RTP payload format for G.722.1
ITU-T G.722.1[ITU.G7221] uses 20 ms frames and a sampling rate clock
of 16 kHz or 32kHz, so the RTP[RFC3550] timestamp MUST be in units of
1/16000 or 1/32000 of a second. The RTP payload for G.722.1 has the
format shown in Figure 1. No additional header specific to this
payload format is required.
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 [3] |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| |
+ one or more frames of G.722.1 |
| .... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: RTP payload for G.722.1
The encoding and decoding algorithm can change the bit rate at any
20ms frame boundary, but no bit rate change notification is provided
in-band with the bit stream. Therefore, a separate out-of-band
method is REQUIRED to indicate the bit rate (see section 6 for an
example of signaling bit rate information using SDP). For the
payload format specified here, the bit rate MUST remain constant for
a particular payload type value. An application MAY switch bit rates
from packet to packet by defining two payload type values and
switching between them.
The assignment of an RTP payload type for this new packet format is
outside the scope of this document, and will not be specified here.
It is expected that the RTP profile for a particular class of
applications will assign a payload type for this encoding, or if that
is not done then a payload type in the dynamic range shall be chosen.
The number of bits within a frame is fixed, and within this fixed
frame G.722.1 uses variable length coding (e.g., Huffman coding) to
represent most of the encoded parameters. All variable length codes
are transmitted in order from the left most (most significant - MSB)
bit to the right most (least significant - LSB) bit, see [ITU.G7221]
for more details.
The use of Huffman coding means that it is not possible to identify
the various encoded parameters/fields contained within the bit stream
without first completely decoding the entire frame. For the purposes
of packetizing the bit stream in RTP, it is only necessary to
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consider the sequence of bits as output by the G.722.1 encoder, and
present the same sequence to the decoder. The payload format
described here maintains this sequence.
When operating at 24 kbit/s, 480 bits (60 octets) are produced per
frame. When operating at 32 kbit/s, 640 bits (80 octets) are
produced per frame. When operating at 48 kbit/s, 960 bits (120
octets) are produced per frame. Thus, all three bit rates allow for
octet alignment without the need for padding bits.
Figure 2 illustrates how the G.722.1 bit stream MUST be mapped into
an octet aligned RTP payload.
first bit last bit
transmitted transmitted
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ sequence of bits (480, 640 or 960) generated by the |
| G.722.1 encoder for transmission |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | | | |
| | | ... | |
| | | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ ... +-+-+-+-+-+-+-+-+-+-+-+-+
|MSB... LSB|MSB... LSB| |MSB... LSB|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ ... +-+-+-+-+-+-+-+-+-+-+-+-+
RTP RTP RTP
octet 1 octet 2 octet
60, 80, 120
Figure 2: The G.722.1 encoder bit stream is split into
a sequence of octets (60, 80 or 120 depending on
the bit rate), and each octet is in turn
mapped into an RTP octet.
The ITU-T standardized bit rates for G.722.1 are 24 kbit/s or
32kbit/s at 16 Khz sample rate, and 24 kbit/s, 32 kbit/s or 48 kbit/s
at 32 Khz sample rate. However, the coding algorithm itself has the
capability to run at any user specified bit rate (not just 24, 32 and
48 kbit/s) while maintaining an audio bandwidth of 50 Hz to 14 kHz.
This rate change is accomplished by a linear scaling of the codec
operation, resulting in frames with size in bits equal to 1/50 of the
corresponding bit rate.
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When operating at non-standard rates the payload format MUST follow
the guidelines illustrated in Figure 2. It is RECOMMENDED that
values in the range 16000 to 48000 be used, and that any value MUST
be a multiple of 400 (this maintains octet alignment and does not
then require (undefined) padding bits for each frame if not octet
aligned). For example, a bit rate of 16.4 kbit/s will result in a
frame of size 328 bits or 41 octets which are mapped into RTP per
Figure 2.
3.1 Multiple G.722.1 frames in a RTP packet
More than one G.722.1 frame may be included in a single RTP packet by
a sender.
Senders have the following additional restrictions:
o Sender SHOULD NOT include more G.722.1 frames in a single RTP
packet than will fit in the MTU of the RTP transport protocol.
o All frames contained in a single RTP packet MUST be of the same
length, that is they MUST have the same bit rate (octets per
frame).
o Frames MUST NOT be split between RTP packets.
It is RECOMMENDED that the number of frames contained within an RTP
packet be consistent with the application. For example, in a
telephony application where delay is important, then the fewer frames
per packet the lower the delay, whereas for a delay insensitive
streaming or messaging application, many frames per packet would be
acceptable.
3.2 Computing the number of G.722.1 frames
Information describing the number of frames contained in an RTP
packet is not transmitted as part of the RTP payload. The only way
to determine the number of G.722.1 frames is to count the total
number of octets within the RTP packet, and divide the octet count by
the number of expected octets per frame.
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4. IANA Considerations
This section describes the media types and names associated with this
payload format. The section registers the media types, as per
RFC2048[RFC2048]
4.1 Registration of MIME media type audio/G7221
MIME media type name: audio
MIME subtype name: G7221
Required parameters:
bitrate: the data rate for the audio bit stream. This parameter is
mandatory because the bit rate is not signaled within the G.722.1 bit
stream. At the standard G.722.1 bit rates, the value MUST be either
24000 or 32000 at 16 Khz sample rate, and 24000, 32000 or 48000 at
32 Khz sample rate. If using the non-standard bit rates, then it is
RECOMMENDED that values in the range 16000 to 48000 be used, and that
any value MUST be a multiple of 400 (this maintains octet alignment
and does not then require (undefined) padding bits for each frame if
not octet aligned).
Optional parameters:
ptime: RECOMMENDED duration of each packet in milliseconds.
Encoding considerations:
This type is only defined for transfer via RTP [RFC3550]
Security considerations: See Section 6
Interoperability considerations: Terminals SHOULD offer a media type
at 16 Khz sample rate in order to interoperate with terminals that do
not support the new 32 Khz sample rate.
Published specification: See ITU-T Recommendation G.722.1 for
encoding algorithm details.
Applications which use this media type:
Audio and Video streaming and conferencing applications.
Additional information: none
Person and email address to contact for further information :
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Roni Even: roni.even@polycom.co.il
Intended usage: COMMON
Author:
Roni Even
Change controller:
IETF Audio/Video Transport working group delegated from the IESG.
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5. SDP Parameters
The media types audio/G7221 are mapped to fields in the Session
Description Protocol (SDP)[RFC2327] as follows:
o The media name in the "m=" line of SDP MUST be audio.
o The encoding name in the "a=rtpmap" line of SDP MUST be G7221 (the
MIME subtype).
o The clock rate in the "a=rtpmap" line MUST be 16000 or 32000.
o One optional parameter MUST be included in the "a=fmtp" line of
SDP. One bitrate SHALL be defined for each payload type.
5.1 Usage with the SDP Offer Answer Model
When offering G.722.1 over RTP using SDP in an Offer/Answer
model[RFC3264] the following considerations are necessary.
There are two clock rates defined for the updated G.722.1.
RFC3047[RFC3047] supported only the 16 Khz clock rate. Therefore a
system that wants to use G.722.1 SHOULD offer a payload type with
clock rate of 16000.
An example of an offer that includes a 16000 and 32000 clock rate is:
m=audio 49000 RTP/AVP 121 122
a=rtpmap:121 G7221/16000
a=fmtp:121 bitrate=24000
a=rtpmap:122 G7221/32000
a=fmtp:121 bitrate=48000
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6. Security Considerations
RTP packets using the payload format defined in this specification
are subject to the security considerations discussed in the RTP
specification [3], 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
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.
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7. Changes from RFC 3047
The new draft updates RFC3047 adding the support for the Wideband
audio support defined in the new revision of ITU-T G.722.1.
Other changes
Update the text to be in line with the current rules for RFC.
8. Normative References
[ITU.G7221]
International Telecommunications Union, "Low-complexity
coding at 24 and 32 kbit/s for hands-free operation in
systems with low frame loss", ITU-T Recommendation
G.722.1, 2005.
[RFC2048] Freed, N., Klensin, J., and J. Postel, "Multipurpose
Internet Mail Extensions (MIME) Part Four: Registration
Procedures", BCP 13, RFC 2048, November 1996.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2327] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
[RFC3047] Luthi, P., "RTP Payload Format for ITU-T Recommendation
G.722.1", RFC 3047, January 2001.
[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.
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Authors' Addresses
Patrick Luthi
Tandberg
Philip Pedersens vei 22
Lysaker 1366
Norway
Email: patrick.luthi@tandberg.no
Roni Even (editor)
Polycom
94 Derech Em Hamoshavot
Petach Tikva 49130
Israel
Email: roni.even@polycom.co.il
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