draft-ietf-avt-rtp-isac-01.txt   draft-ietf-avt-rtp-isac-02.txt 
Network Working Group T. le Grand Network Working Group T. le Grand
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track P. Jones Intended status: Standards Track P. Jones
Expires: October 19, 2012 P. Huart Expires: April 21, 2013 P. Huart
Cisco Systems Cisco Systems
T. Shabestary
H. Alvestrand, Ed. H. Alvestrand, Ed.
Google Google
April 17, 2012 October 18, 2012
RTP Payload Format for the iSAC Codec RTP Payload Format for the iSAC Codec
draft-ietf-avt-rtp-isac-01 draft-ietf-avt-rtp-isac-02
Abstract Abstract
iSAC is a proprietary wideband speech and audio codec developed by iSAC is a proprietary wideband speech and audio codec developed by
Global IP Solutions, suitable for use in Voice over IP applications. Global IP Solutions (now part of Google), suitable for use in Voice
This document describes the payload format for iSAC generated bit over IP applications. This document describes the payload format for
streams within a Real-Time Protocol (RTP) packet. Also included here iSAC generated bit streams within a Real-Time Protocol (RTP) packet.
are the necessary details for the use of iSAC with the Session Also included here are the necessary details for the use of iSAC with
Description Protocol (SDP). the Session Description Protocol (SDP).
Requirements Language Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [1]. document are to be interpreted as described in RFC 2119 [1].
Status of this Memo Status of this Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
skipping to change at page 1, line 45 skipping to change at page 1, line 46
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on October 19, 2012. This Internet-Draft will expire on April 21, 2013.
Copyright Notice Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
skipping to change at page 2, line 18 skipping to change at page 2, line 19
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. iSAC Codec Description . . . . . . . . . . . . . . . . . . . . 3 2. iSAC Codec Description . . . . . . . . . . . . . . . . . . . . 3
3. RTP Payload Format . . . . . . . . . . . . . . . . . . . . . . 4 3. RTP Payload Format . . . . . . . . . . . . . . . . . . . . . . 5
3.1. iSAC Payload Format . . . . . . . . . . . . . . . . . . . . 4 3.1. iSAC Wideband Payload Format . . . . . . . . . . . . . . . 5
3.2. Payload Header . . . . . . . . . . . . . . . . . . . . . . 5 3.2. Payload Header . . . . . . . . . . . . . . . . . . . . . . 6
3.3. Encoded Speech Data . . . . . . . . . . . . . . . . . . . . 5 3.3. Encoded Speech Data . . . . . . . . . . . . . . . . . . . 6
3.4. Multiple iSAC frames in an RTP packet . . . . . . . . . . . 6 3.4. iSAC Superwideband Payload Format . . . . . . . . . . . . 7
4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 6 3.5. Encoded Upper-band Speech Data . . . . . . . . . . . . . . 8
4.1. Media Type registration of iSAC . . . . . . . . . . . . . . 7 3.6. Padding . . . . . . . . . . . . . . . . . . . . . . . . . 8
5. Mapping to SDP Parameters . . . . . . . . . . . . . . . . . . . 7 3.7. Multiple iSAC frames in an RTP packet . . . . . . . . . . 9
5.1. Example Initial Target Bit Rate . . . . . . . . . . . . . . 8 4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
5.2. Example Max Bit Rate . . . . . . . . . . . . . . . . . . . 8 5. Mapping to SDP Parameters . . . . . . . . . . . . . . . . . . 10
6. Security Considerations . . . . . . . . . . . . . . . . . . . . 9 5.1. Example Initial Target Bit Rate . . . . . . . . . . . . . 11
7. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . 9 5.2. Example Max Bit Rate . . . . . . . . . . . . . . . . . . . 11
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 9 6. Security Considerations . . . . . . . . . . . . . . . . . . . 12
8.1. Normative References . . . . . . . . . . . . . . . . . . . 9 7. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 12
8.2. Informative References . . . . . . . . . . . . . . . . . . 9 8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 12
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 9 8.1. Normative References . . . . . . . . . . . . . . . . . . . 12
8.2. Informative References . . . . . . . . . . . . . . . . . . 12
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 12
1. Introduction 1. Introduction
This document gives a general description of the iSAC wideband speech This document gives a general description of the iSAC wideband speech
codec and specifies the iSAC payload format for usage in RTP packets. codec and specifies the iSAC payload format for usage in RTP packets.
Also included here are the necessary details for the use of iSAC with Also included here are the necessary details for the use of iSAC with
the Session Description Protocol (SDP). the Session Description Protocol (SDP).
2. iSAC Codec Description 2. iSAC Codec Description
The iSAC codec is an adaptive wideband speech and audio codec that The iSAC codec is an adaptive wideband/superwideband speech and audio
operates with short delay, making it suitable for high quality real codec that operates with short delay, making it suitable for high
time communication. It is specially designed to deliver wideband quality real time communication. It is specially designed to deliver
speech quality in both low and medium bit rate applications. It also wideband speech quality in both low and medium bit rate applications.
handles non-speech audio well, such as music and background noise It also handles non-speech audio well, such as music and background
[5]. noise [5].
The iSAC codec compresses speech frames of 16 kHz, 16-bit sampled The iSAC codec compresses speech frames of 16 kHz, 16-bit sampled
input speech, each frame containing 30 or 60 ms of speech. input speech, each frame containing 30 or 60 ms of speech. It also
has a superwideband mode which allows a 32 kHz sampling rate. In
super-wideband mode the input signal is split into wideband (0-8 kHz)
and upper (8-16 kHz) signal. Each sub-band is encoded independently,
and their associated payloads concatenated, c.f. Figure 2, to
construct the overall iSAC super-wideband RTP payload. Note that the
same encoder/decoder is used for the wideband part for both wideband
and super-wideband modes.
The codec runs in one of two different modes called channel-adaptive The codec runs in one of two different modes called channel-adaptive
mode and channel-independent mode. In both modes iSAC is aiming at a mode and channel-independent mode. In both modes iSAC is aiming at a
target bit rate, which is neither the average nor the maximum bit target bit rate, which is neither the average nor the maximum bit
rate that will be reach by iSAC, but corresponds to the average bit rate that will be reach by iSAC, but corresponds to the average bit
rate during peaks in speech activity. The bit rate will sometimes rate during peaks in speech activity. The bit rate will sometimes
exceed the target bit rate, but most of the time will be below. The exceed the target bit rate, but most of the time will be below. The
average bit rate obtained is on average about a factor of 1.4 times average bit rate obtained is on average about a factor of 1.2 times
lower than the target bit rate. lower than the target bit rate on continuous speech, and will be
lower on speech with pauses.
In channel-adaptive mode the target bit rate is adapted to give a bit In channel-adaptive mode the target bit rate is adapted to give a bit
rate corresponding to the available bandwidth on the channel. The rate corresponding to the available bandwidth on the channel. The
available bandwidth is constantly estimated at the receiving iSAC and available bandwidth is continuously estimated at the receiving iSAC
signaled in-band in the iSAC bit stream. Even at dial-up modem data and signaled in-band in the iSAC bit stream. Even at dial-up modem
rates (including IP, UDP, and RTP overhead) iSAC delivers high data rates (including IP, UDP, and RTP overhead) iSAC delivers high
quality by automatically adjusting transmission rates to give the quality by automatically adjusting transmission rates to give the
best possible listening experience over the available bandwidth. The best possible listening experience over the available bandwidth. The
default initial target bit rate is 20000 bits per second in channel- default initial target bit rate is 20000 bits per second in channel-
adaptive mode. adaptive mode.
In channel-independent mode a target bit rate has to be provided to In channel-independent mode a target bit rate has to be provided to
iSAC prior to encoding. iSAC prior to encoding; the target bit rate can be changed over the
time of the call.
After encoding the speech signal the iSAC coder uses lossless coding After encoding the speech signal the iSAC coder uses lossless coding
to further reduce the size of each packet, and hence the total bit to further reduce the size of each packet, and hence the total bit
rate used. rate used.
The adaptation and the lossless coding described above both result in The adaptation and the lossless coding described above both result in
a variation of packet size, depending both of the nature of speech a variation of packet size, depending both of the nature of speech
and the available bandwidth. Therefore the iSAC codec operates at and the available bandwidth. Therefore, the iSAC codec, in wideband
transmission rates from about 10 kbps to about 32 kbps. mode, operates at transmission rates from about 10 kbps to about 32
kbps. In super-wideband mode, the transmission rate is in the range
of 10 kbps to 56 kbps. If operating in super-wideband mode, the iSAC
codec automatically adjusts the effective encoded audio bandwidth for
the best experience.
Bit Rate | 10 - 32 | 32 - 38 | 38 - 45 | 45 - 50 | 50 - 56
[kbps] | | | | |
----------+----------+------------+----------------------+---------
Effective | | 0 - 8 | 0 - 12 | 0 - 12 | 0 - 16
Bandwidth | 0 - 8 kHz| operating | | operating |
[kHz] | | at 32 kbps | | at 45 kbps |
The main characteristics can be summarized as follows: The main characteristics can be summarized as follows:
o Wideband, 16 kHz, speech and audio codec o Wideband or superwideband, 16 kHz or 32 kHz respectively, speech
and audio codec
o Variable bit rate, which depends on the input signal o Variable bit rate, which depends on the input signal
o Adaptive rate with two modes: channel-adaptive or channel- o Adaptive rate with two modes: channel-adaptive or channel-
independent mode independent mode
o Bit rate range from around 10 kbps to 32 kbps o Bit rate range from around 10 kbps to 32 kbps when operating on
wideband input. For input audio sampled at 32 kHz, the bit rate
range 10 kbps to 56 kbps.
o Operates on 30 or 60 ms of speech o Operates on 30 or 60 ms of speech for wideband inputs, and only 30
ms for super-wideband inputs.
o In super-wideband mode, depending on the target bit rate, the
effective bandwidth is adjusted for the optimal experience.
3. RTP Payload Format 3. RTP Payload Format
The iSAC codec uses a sampling rate clock of 16 kHz, so the RTP The iSAC codec in wideband mode uses a sampling rate clock of 16 kHz,
timestamp MUST be in units of 1/16000 of a second. so the RTP timestamp MUST be in units of 1/16000 of a second. In
super-wideband mode, the iSAC codec uses a sampling rate clock of 32
kHz, so the RTP timestamp MUST be in units of 1/32000 of a second.
The RTP payload for iSAC has the format shown in Figure 1. No The RTP payload for iSAC has the format shown in Figure 1. No
additional header fields specific to this payload format are additional header fields specific to this payload format are
required. For RTP based transportation of iSAC encoded audio, the required. For RTP based transportation of iSAC encoded audio, the
standard RTP header [2] is followed by one payload data block. standard RTP header [2] is followed by one payload data block.
0 1 2 3 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 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 | | RTP Header |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| | | |
+ iSAC Payload Block + + iSAC Payload Block +
| | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: RTP packet format for iSAC Figure 1: RTP packet format for iSAC
skipping to change at page 4, line 40 skipping to change at page 5, line 28
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 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 | | RTP Header |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| | | |
+ iSAC Payload Block + + iSAC Payload Block +
| | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: RTP packet format for iSAC Figure 1: RTP packet format for iSAC
3.1. iSAC Payload Format 3.1. iSAC Wideband Payload Format
The iSAC payload block consists of a payload header and one or two The iSAC payload block consists of a payload header and one or two
encoded 30 ms speech frames. The iSAC payload is generated in the encoded 30 ms speech frames. The iSAC payload is generated in the
following manner: following manner:
o Parameters representing one or two 30 ms frames of speech data are o Parameters representing one or two 30 ms frames of speech data are
determined by the encoder. The parameters are quantized to determined by the encoder. The parameters are quantized to
generate encoded data corresponding to the one or two speech generate encoded data corresponding to the one or two speech
frames. The length of the encoded data is variable and depends on frames. The length of the encoded data is variable and depends on
the signal characteristics and the target bit rate. the signal characteristics and the target bit rate.
skipping to change at page 5, line 17 skipping to change at page 5, line 52
o Lossless coding is applied to the complete iSAC payload block, o Lossless coding is applied to the complete iSAC payload block,
including payload header, to generate a compressed payload. The including payload header, to generate a compressed payload. The
length depends on the length of the data generated to represent length depends on the length of the data generated to represent
the speech and the effectiveness of the lossless coding. the speech and the effectiveness of the lossless coding.
No part of the payload header or the encoded speech data can be No part of the payload header or the encoded speech data can be
retrieved without partly or fully decoding the packet. retrieved without partly or fully decoding the packet.
The following figure shows an iSAC payload block containing 60 ms of The following figure shows an iSAC payload block containing 60 ms of
encoded speech data: encoded speech data.
+--------+--------+--------+--------+--------+--------+--------+
|Payload | 30 ms Encoded | 30 ms Encoded | +--------+--------+--------+--------+--------+--------+------+
|Header | Speech Data | Speech Data | |Payload | 30 ms Encoded | 30 ms Encoded |
+--------+--------+--------+--------+--------+--------+--------+ |Header | Speech Data | Speech Data |
+--------+--------+--------+--------+--------+--------+------+
Figure 2: Payload format for iSAC Figure 2: Payload format for iSAC
3.2. Payload Header 3.2. Payload Header
The payload header holds information for the receiver about the The payload header holds information for the receiver about the
available bandwidth (BEI), and the length of the speech data in the available bandwidth, in the form of a Bandwidth Estimation Index
current payload (FL). The header has the format defined in Figure 3. (BEI), and the length of the speech data in the current payload
(frame length, FL). The header has the format defined in Figure 3.
Note that the size of the header can vary due to the lossless Note that the size of the header can vary due to the lossless
encoding described in section 2 and in section 3.1. Also note that encoding described in section 2 and in section 3.1. Also note that
the BEI is always estimated and transmitted, even if iSAC runs in the BEI is always estimated and transmitted, even if iSAC runs in
channel-independent mode. channel-independent mode.
+-+-+-+-+-+-+ +-+-+-+-+-+-+
| BEI | FL | | BEI | FL |
+-+-+-+-+-+-+ +-+-+-+-+-+-+
Figure 3: Payload Header Figure 3: Payload Header
o BEI: Bandwidth Estimation Index. The bandwidth estimate is o BEI: Bandwidth Estimation Index. The bandwidth estimate is
quantized into one out of 24 values. Valid values are 0 to 23. quantized into one out of 24 values. Valid values are 0 to 23.
o FL: The length of the speech data (Frame Length) present in the o FL: The length of the speech data (Frame Length) present in the
payload, given in number of speech samples. Valid frame lengths payload, given in number of speech samples. Valid frame lengths
skipping to change at page 5, line 51 skipping to change at page 6, line 40
o FL: The length of the speech data (Frame Length) present in the o FL: The length of the speech data (Frame Length) present in the
payload, given in number of speech samples. Valid frame lengths payload, given in number of speech samples. Valid frame lengths
are 480 (30 ms) and 960 (60 ms) samples. are 480 (30 ms) and 960 (60 ms) samples.
3.3. Encoded Speech Data 3.3. Encoded Speech Data
The iSAC encoded speech data consist of parameters representing one The iSAC encoded speech data consist of parameters representing one
or two frames of 30 ms speech. The length of the speech data is or two frames of 30 ms speech. The length of the speech data is
signaled in the header (in number of samples), and the length may signaled in the header (in number of samples), and the length may
change at any time during a session. In channel-adaptive mode the change at any time during a session. In channel-adaptive mode the
length is changed to best utilize the available bandwidth. length is changed to best utilize the available bandwidth, and extra
padding is added to some packets as a bandwidth probe.
The iSAC payload is padded to whole octets, and has a variable length The iSAC payload is padded to whole octets, and has a variable length
depending on the input source signal, number of 30 ms speech frames, depending on the input source signal, number of 30 ms speech frames,
and target bit rate. and target bit rate.
The number of octets used to describe one frame of 30 ms speech The number of octets used to describe one frame of 30 ms speech
typically varies from around 50 to around 120 octets. For the case typically varies from around 50 to around 120 octets. For the case
of 60 ms speech (two 30 ms speech frames), the number of octets of 60 ms speech (two 30 ms speech frames), the number of octets
varies from around 100 to around 240 octets. The absolute maximum varies from around 100 to around 240 octets. The absolute maximum
allowed payload length is 400 octets. The user can choose to lower allowed payload length is 400 octets. The user can choose to lower
the maximum allowed payload length. Minimum value is 100 octets. It the maximum allowed payload length. Minimum value is 100 octets. It
is possible for the user to choose a maximum bit rate instead of a is possible for the user to choose a maximum bit rate (averaged over
maximum payload length. The maximum payload length is then dependent a frame) instead of a maximum payload length. The maximum payload
on the length of the speech data represented in the payload (30 or 60 length is then dependent on the length of the speech data represented
ms). Possible maximum rates are in the range of 32000 to 53400 bits in the payload (30 or 60 ms). Possible maximum rates are in the
per second. range of 32000 to 53400 bits per second.
The sensitivity to bit errors is equal for all bits in the payload. The sensitivity to bit errors is equal for all bits in the payload.
3.4. Multiple iSAC frames in an RTP packet 3.4. iSAC Superwideband Payload Format
In super-wideband mode, payloads associated with each sub-band
(wideband 0-8 kHz and upper-band 8-16 kHz) are constructed
independently and concatenated as depicted in Figure 2. Note that in
super-wideband mode only one 30 ms frame is encoded in each payload.
+--------------------------------+---+------------------------+-----+
| Payload +30 ms Encoded wideband|LEN|30 ms Encoded upper-band| CRC |
| Header speech data | |speech data |check|
+--------------------------------+---+------------------------+-----+
|<--- CRC checked data ->|
Figure 4: Super-Wideband payload format
Payloads of wideband and upper-band are encoded independently,
allowing the encoder to simply concatenate two payloads to construct
one iSAC super-wideband payload. The RTP payload of the iSAC super-
wideband codec starts with the payload of the wideband part, which is
padded to whole octets, followed by one byte (LEN in Figure 4)
representing the length of the remaining sequence, payload of the
upper-band plus 4 bytes for CRC sequence.
If LEN_UB denotes the length of the upper-band payload, then LEN = 1
+ LEN_UB + 4. This value should not exceed 255, otherwise upper-band
payload is omitted.
The CRC check is added to distinguish between upper-band payload and
random bit-stream padding that can be added for probing available
network bandwidth.
At the receive side, a super-wideband payload is first given to the
wideband decoder. The wideband decoder decodes as many parameters as
required to uniquely reproduce the encoded wideband audio. The next
byte in the payload should hold the value of LEN. This provides a
sanity check that the decoding process has not failed. Thereafter,
the receiver runs a CRC check over the upper-band payload and
compares the results with the last 4 bytes in the packet.
If the computed CRC and the last four bytes of the payload don't
match, the remaining bits are assumed to be added for probing the
network. Hence, the upper-band signal is replaced by zeros and
combined with the wideband signal to generate the super-wideband
signal.
If the two CRCs match, then the upper-band payload is given to the
upper-band decoder. Thereby, the output of the upper-band decoder is
combined with the wide-band decoded audio to generate the super-
wideband signal.
It might be that for a given packet, the wideband decoder uses all
the given payload. This can be the case when a super-wideband
encoder is operating at low rates and has adjusted the effective
bandwidth to wideband. In this case, the decoder inserts zeros as
the reconstructed upper-band and combines both bands to reproduce the
super-wideband signal.
3.5. Encoded Upper-band Speech Data
The iSAC encoded upper-band speech data consists of parameters
representing one frame of 30 ms speech. Depending on the target rate
the upper-band encoder might choose to only encode the sub-band of 8
kHz to 12 kHz. This is signaled inband to the receiver.
3.6. Padding
Padding, which consists of randomly generated bits, may be added at
the end of the payload in both wideband and superwideband modes. It
can be used by the sender for bandwidth probing, and is always
ignored by the receiver.
In wideband mode, padding simply follows the payload, preceded by a
length field.
+----------+---+--------+
| Wideband |LEN|Padding |
| payload | | |
+----------+---+--------+
Figure 5: Wideband payload format with padding.
LEN is the length of the padding in bytes + 1: LEN = LEN_PAD + 1
In superwideband mode, the format of a packet with padding looks like
the following.
+----------+---+-------------+--+--------+-----+
| Wideband |LEN|Upper-band |L2|Padding |CRC |
| payload | |speech data | | |check|
+----------+---+-------------+--+--------+-----+
|<-- CRC checked data --->|
Figure 6: Super-Wideband payload format
LEN is 1 + LEN_UB + 1 + LEN_PAD + 4, where LEN_UB is the length of
the upper-band speech data in bytes, and LEN_PAD is the length of the
padding in bytes.
L2 is LEN_PAD + 1.
The CRC check runs over the upper-band speech data, L2 and the
padding.
3.7. Multiple iSAC frames in an RTP packet
More than one iSAC payload block MUST NOT be included in an RTP More than one iSAC payload block MUST NOT be included in an RTP
packet by a sender. packet by a sender.
Further, iSAC payload blocks MUST NOT be split between RTP packets. Further, iSAC payload blocks MUST NOT be split between RTP packets.
4. IANA Considerations 4. IANA Considerations
This document defines the iSAC media type. This document defines the iSAC media type, and requests IANA to
register it.
Media type name: audio Media type name: audio
Media subtype: isac Media subtype: isac
Required parameters: None Required parameters: None
Optional parameters: Optional parameters:
* ibitrate: The parameter indicates the upper bound of the * ibitrate: The parameter indicates the upper bound of the
skipping to change at page 7, line 29 skipping to change at page 10, line 29
This media type is suitable for use in numerous applications This media type is suitable for use in numerous applications
needing to transport encoded voice or other audio. Some examples needing to transport encoded voice or other audio. Some examples
include Voice over IP, Streaming Media, Voice Messaging, and include Voice over IP, Streaming Media, Voice Messaging, and
Conferencing. Conferencing.
Additional information: None Additional information: None
Intended usage: COMMON Intended usage: COMMON
Other Information/General Comment: Other Information/General Comment:
iSAC is a proprietary speech and audio codec owned by Global IP iSAC is a proprietary speech and audio codec owned by Google. The
Solutions. The codec operates on 30 or 60 ms speech frames at a codec operates on 30 or 60 ms speech frames at a sampling rate
sampling rate clock of 16 kHz. clock of 16 kHz or 32 kHz.
Person to contact for further information: Person to contact for further information:
Tina le Grand [tlegrand@google.com] Tina le Grand [tlegrand@google.com]
Restrictions on usage: Restrictions on usage:
This media type depends on RTP framing, and hence is only defined This media type depends on RTP framing, and hence is only defined
for transfer via RTP [2] Transport within other framing protocols for transfer via RTP [2]. Transport within other framing
is not defined at this time. protocols is not defined at this time.
Change controller: Change controller:
IETF Audio/Video Transport working group delegated from the IESG. IETF Audio/Video Transport working group delegated from the IESG.
Note to the RFC Editor / IANA: Please replace "RFC XXXX" above with Note to the RFC Editor / IANA: Please replace "RFC XXXX" above with
the number of this RFC when published, and remove this note. the number of this RFC when published, and remove this note.
4.1. Media Type registration of iSAC
5. Mapping to SDP Parameters 5. Mapping to SDP Parameters
The information carried in the media type specification has a The information carried in the media type specification has a
specific mapping to fields in the Session Description Protocol (SDP) specific mapping to fields in the Session Description Protocol (SDP)
[3], which is commonly used to describe RTP sessions. When SDP is [3], which is commonly used to describe RTP sessions. When SDP is
used to specify sessions employing the iSAC codec, the mapping is as used to specify sessions employing the iSAC codec, the mapping is as
follows: follows:
o The media type ("audio") goes in SDP "m=" as the media name. o The media type ("audio") goes in SDP "m=" as the media name.
skipping to change at page 8, line 32 skipping to change at page 11, line 30
those in the SDP answer. For both ibitrate and maxbitrate it is those in the SDP answer. For both ibitrate and maxbitrate it is
legal for the answer to contain a value that is different than what legal for the answer to contain a value that is different than what
is provided in an offer. The parameter may be present in the answer, is provided in an offer. The parameter may be present in the answer,
even if absent in the offer. even if absent in the offer.
When conveying information by SDP, the encoding name SHALL be "isac" When conveying information by SDP, the encoding name SHALL be "isac"
(the same as the media subtype). (the same as the media subtype).
5.1. Example Initial Target Bit Rate 5.1. Example Initial Target Bit Rate
The offer indicates that it wishes to receive a bitstream with an The offer indicates that it wishes to receive a wideband bitstream
initial target rate of 20000 bits per second. The remote party MAY with an initial target rate of 20000 bits per second. The remote
change its initial target rate to the requested value. party MAY change its initial target rate to the requested value.
m=audio 10000 RTP/AVP 98 m=audio 10000 RTP/AVP 98
a=rtpmap: 98 isac/16000 a=rtpmap: 98 isac/16000
a=fmtp:98 ibitrate=20000 a=fmtp:98 ibitrate=20000
5.2. Example Max Bit Rate 5.2. Example Max Bit Rate
The offer indicates that it wishes to receive a bitstream with an The offer indicates that it wishes to receive a superwideband
initial target rate of 20000 bits per second, and a maximum bit rate bitstream with an initial target rate of 20000 bits per second, and a
of 45000 bits per second. The remote party MAY change its initial maximum bit rate of 45000 bits per second. The remote party MAY
target rate and SHOULD NOT transmit at a higher rate than 45000. change its initial target rate and SHOULD NOT transmit at a higher
rate than 45000.
m=audio 10000 RTP/AVP 98 m=audio 10000 RTP/AVP 98
a=rtpmap: 98 isac/16000 a=rtpmap: 98 isac/32000
a=fmtp:98 ibitrate=20000;maxrate=45000 a=fmtp:98 ibitrate=20000;maxbitrate=45000
6. Security Considerations 6. Security Considerations
RTP packets using the payload format defined in this specification RTP packets using the payload format defined in this specification
are subject to the general security considerations discussed in RFC are subject to the general security considerations discussed in RFC
3550 8.1. 3550 section 8.1.
As this format transports encoded speech, the main security issues As this format transports encoded speech, the main security issues
include confidentiality and authentication of the speech itself. The include confidentiality and authentication of the speech itself. The
payload format itself does not have any built-in security mechanisms. payload format itself does not have any built-in security mechanisms.
External mechanisms, such as SRTP [4], MAY be used. External mechanisms, such as SRTP [4], MAY be used.
7. Acknowledgments 7. Acknowledgments
This document was originally prepared using 2-Word-v2.0.template.dot. This document was originally prepared using 2-Word-v2.0.template.dot.
skipping to change at page 9, line 42 skipping to change at page 12, line 42
[3] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session [3] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006. Description Protocol", RFC 4566, July 2006.
[4] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. [4] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004. RFC 3711, March 2004.
8.2. Informative References 8.2. Informative References
[5] "iSAC datasheet at Golbal IP Solutions website [5] GIPS / Google, "iSAC reference implementation".
http://www.gipscorp.com/files/english/datasheets/iSAC.pdf".
Available at http://code.google.com/p/webrtc/source - directory
src/modules/audio_coding/codecs/isac
Authors' Addresses Authors' Addresses
Tina le Grand Tina le Grand
Google Google
Kungsbron 2 Kungsbron 2
Stockholm, 11122 Stockholm, 11122
Sweden Sweden
Paul E. Jones Paul E. Jones
skipping to change at page 10, line 35 skipping to change at page 13, line 35
Cisco Systems Cisco Systems
400, Avenue Roumanille, Batiment T3 400, Avenue Roumanille, Batiment T3
Biot - Sophia Antipolis, 06410 Biot - Sophia Antipolis, 06410
France France
Phone: +33 4 9723 2643 Phone: +33 4 9723 2643
Fax: Fax:
Email: phuart@cisco.com Email: phuart@cisco.com
URI: URI:
Turaj Zakizadeh Shabestary
Google
1950 Charleston Road
Mountain View, CA 94043
USA
Phone:
Fax:
Email: turajs@google.com
URI:
Harald Alvestrand (editor) Harald Alvestrand (editor)
Google Google
Kungsbron 2 Kungsbron 2
Stockholm, 11122 Stockholm, 11122
Sweden Sweden
Phone: Phone:
Fax: Fax:
Email: hta@google.com Email: hta@google.com
URI: URI:
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