Audio Video Transport WG Sassan Ahmadi
INTERNET-DRAFT Nokia Inc.
Category: Standards Track May 17, 2004
Expires: November 17, 2004
Real-Time Transport Protocol (RTP) Payload and File Storage
Formats for the Variable-Rate Multimode Wideband (VMR-WB)
Audio Codec
<draft-ahmadi-avt-rtp-vmr-wb-02.txt>
Status of this Memo
This document is an Internet-Draft and is in full conformance
with all provisions of Section 10 of RFC 2026
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Copyright Notice
Copyright (C) The Internet Society (2004). All Rights
Reserved.
Abstract
This document specifies a real-time transport protocol (RTP)
payload format to be used for the Variable-Rate Multimode
Wideband (VMR-WB) speech codec. The payload format is
designed to be able to interoperate with existing VMR-WB
transport formats on non-IP networks. In addition, a file
format is specified for transport of VMR-WB speech data in
storage mode applications such as email. A MIME type
registration is included, for VMR-WB, specifying use of
both the RTP payload and the storage formats
VMR-WB is a variable-rate multimode wideband speech codec
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that has a number of operating modes, one of which is
interoperable with AMR-WB (i.e., RFC 3267) audio codec at
certain rates. Therefore, provisions have been made in
this draft to facilitate and simplify data packet exchange
between VMR-WB and AMR-WB in the interoperable mode with no
transcoding function involved.
Table of Contents
1.Introduction.................................................3
2.Conventions and Acronyms.....................................3
3.The Variable-Rate Multimode Wideband (VMR-WB) Speech Codec...4
3.1. Narrowband Speech Processing...........................5
3.2. Continuous vs. Discontinuous Transmission..............5
3.3. Support for Multi-Channel Session......................6
4. Robustness against Packet Loss..............................6
4.1. Forward Error Correction (FEC).........................6
4.2. Frame Interleaving and Multi-Frame Encapsulation.......7
5. VMR-WB Voice over IP scenarios..............................8
5.1. IP Terminal to IP Terminal.............................8
5.2 IP Terminal to GW to IP Terminal.......................8
5.3. GW to IP Terminal......................................9
5.4. GW to GW (Between VMR-WB and AMR-WB Enabled Terminals)10
5.5. GW to GW (Between two VMR-WB Enabled Terminals).......11
6. VMR-WB RTP Payload Formats.................................11
6.1. RTP Header Usage.............................. .......13
6.2. Header-Free Payload Format............................12
6.3. Octet-Aligned Payload Format..........................14
6.3.1. Payload Structure................................14
6.3.2. The Payload Header...............................14
6.3.3. The Payload Table of Contents....................17
6.3.4. Speech Data......................................19
6.3.5. Payload Example..................................20
Basic Single Channel Payload Carrying Multiple Frames
6.4. Implementation Considerations.........................20
7. VMR-WB Storage Format......................................20
7.1. Single Channel Header.................................21
7.2. Multi-Channel Header..................................21
7.3. Speech Frames.........................................22
8. Congestion Control.........................................23
9. Security Considerations....................................24
9.1. Confidentiality.......................................24
9.2. Authentication........................................25
9.3. Decoding Validation and Provision for Lost or Late
Packets...............................................25
10. Payload Format Parameters.................................25
10.1. VMR-WB MIME Registration.............................26
10.2. Mapping MIME Parameters into SDP.....................28
10.3. Offer-Answer Model Considerations....................29
11. IANA Considerations.......................................30
12. Acknowledgements..........................................30
References....................................................30
Normative References.......................................30
Informative References.....................................30
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Author's Address..............................................31
Full Copyright Statement......................................31
1. Introduction
This document specifies the payload format for packetization
of VMR-WB encoded speech signals into the Real-time Transport
Protocol (RTP) [3]. The VMR-WB payload formats support
transmission of single and multiple channels, frame
interleaving, multiple frames per payload, header-free
payload, the use of mode switching, and interoperation with
existing VMR-WB transport formats on non-IP networks, as
described in Section 3.
The payload format itself is specified in Section 6. A
related file format is specified in Section 7 for transport
of VMR-WB speech data in storage mode applications such as
email. In Section 10, a MIME type registration for VMR-WB is
provided.
Since VMR-WB is interoperable with AMR-WB at certain rates,
an attempt has been made throughout this document to maximize
the similarities with RFC 3267 while optimizing the payload
and storage formats for the non-interoperable modes of the
VMR-WB codec.
2. Conventions and Acronyms
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 [2].
The following acronyms are used in this document:
3GPP2 - The Third Generation Partnership Project 2
CDMA - Code Division Multiple Access
WCDMA - Wideband Code Division Multiple Access
GSM - Global System for Mobile Communications
AMR-WB - Adaptive Multi-Rate Wideband Codec
VMR-WB - Variable-Rate Multimode Wideband Codec
CMR - Codec Mode Request
GW - Gateway
DTX - Discontinuous Transmission
FEC - Forward Error Correction
SID - Silence Descriptor
TrFO - Transcoder-Free Operation
UDP - User Datagram Protocol
RTP - Real-Time Transfer Protocol
RTCP - Real-Time Control Protocol
MIME - Multipurpose Internet Mail Extension
SDP - Session Description Protocol
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SIP - Session Initiation Protocol
The term "frame-block" is used in this document to describe
the time-synchronized set of speech frames in a multi-channel
VMR-WB session. In particular, in an N-channel session, a
frame-block will contain N speech frames, one from each of
the channels, and all N speech frames represent exactly the
same time period.
3. The Variable-Rate Multimode Wideband (VMR-WB) Speech Codec
VMR-WB is the wideband speech-coding standard developed by
Third Generation Partnership Project 2 (3GPP2) for
encoding/decoding wideband/narrowband speech content in
multimedia services in 3G CDMA cellular systems. VMR-WB is a
source-controlled variable-rate multimode wideband speech
codec. It has a number of operating modes, where each mode is
a tradeoff between voice quality and average data rate. The
operating mode in VMR-WB is chosen based on the traffic
condition of the network and the desired quality of
service [1]. The desired average data rate (ADR) in each mode
is obtained by encoding speech frames at different rates
compliant with CDMA Rate-Set II depending on the
instantaneous characteristics of input speech and the
maximum and minimum rate constraints imposed by the network
operator. While VMR-WB is a native CDMA codec complying with
all CDMA system requirements, it is further interoperable
with AMR-WB [4] at 12.65, 8.85, and 6.60 kbps. This is due to
the fact that VMR-WB and AMR-WB share the
same core technology. This feature enables Transcoder Free
(TrFO) interconnections between VMR-WB and AMR-WB across
different wireless/wireline systems (e.g., GSM/WCDMA and
CDMA2000) without use of unnecessary complex media format
conversion.
VMR-WB is able to transition between various modes with no
degradation in voice quality that is attributable to the mode
switching itself. The operation mode of the VMR-WB encoder
may be switched seamlessly without prior knowledge of the
decoder. Any non-interoperable mode (i.e., mode 0, 1, or 2)
can be chosen depending on the traffic conditions (e.g.,
network congestion) and the desired quality of service.
While in the interoperable mode (i.e., VMR-WB mode 3), mode
switching is not allowed. There is only one AMR-WB
interoperable mode in VMR-WB. Since AMR-WB codec depending on
channel conditions may request a mode change, in-band data
included in VMR-WB frame structure (see Section 8 of [1] for
more details), is used during an interoperable
interconnection to switch between AMR-WB codec modes 0, 1, or
2.
As mentioned earlier, VMR-WB is compliant with CDMA Rate-Set
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II (see Section 2 of [1]) with the permissible encoding rates
shown in Table 1.
+--------------+-------------------+-----------------+
| Frame Type | Bits per Packet | Encoding Rate |
| | (Frame Size) | (kbps) |
+--------------+-------------------+-----------------+
| Full-Rate | 266 | 13.3 |
| Half-Rate | 124 | 7.2 |
| Quarter-Rate | 54 | 2.7 |
| Eighth-Rate | 20 | 1.0 |
| Blank | 0 | - |
| Erasure | 0 | - |
+--------------+-------------------+-----------------+
Table 1: CDMA Rate-Set II frame types and their associated
encoding rates
VMR-WB is robust to high percentage of packet loss and
packets with corrupted rate information. The reception of
an Erasure (SPEECH_LOST) frame type at decoder invokes the built-in
frame error concealment mechanism. The built-in frame error
concealment mechanism in VMR-WB conceals the effect of lost
packets by exploiting in-band data and the information
available in the previous frames.
3.1. Narrowband Speech Processing
VMR-WB has the capability to operate with 8000 Hz sampled
input/output speech signals in all modes of operation [1].
Mode switching can be utilized to change the mode of
operation while processing narrowband speech signals.
However, during a session, transition between narrowband and
wideband processing is not RECOMMENDED due to different
timestamps and other likely synchronization problems.
3.2. Continuous vs. Discontinuous Transmission
The circuit-switched operation of VMR-WB within a CDMA
network requires continuous transmission of the speech data
during a conversation. The intrinsic source-controlled
variable-rate feature of the CDMA speech codecs is required
for optimal operation of the CDMA system and interference
control. However, VMR-WB has the capability to operate in a
discontinuous transmission mode for some packet-switched
applications over IP networks, where the number of
transmitted bits and packets during silence period are
reduced to a minimum. The VMR-WB DTX operation is similar to
that of AMR-WB [4,12].
4. Support for Multi-Channel Session
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Both the octet-aligned RTP payload format and the storage
format defined in this document support multi-channel audio
content (e.g., a stereophonic speech session).
Although VMR-WB codec itself does not support encoding of
multi-channel audio content into a single bit stream, it can
be used to separately encode and decode each of the
individual channels.
To transport (or store) the separately encoded multi-channel
content, the speech frames for all channels that are framed
and encoded for the same 20 ms periods are logically
collected in a frame-block.
At the session setup, out-of-band signaling must be used to
indicate the number of channels in the session and the order
of the speech frames from different channels in each frame-
block. When using SDP for signaling, the number of
channels is specified in the rtpmap attribute and the order
of channels carried in each frame-block is implied by the
number of channels as specified in Section 4.1 in [10].
4. Robustness against Packet Loss
The octet-aligned payload format, described in this document,
supports several features including forward error correction
(FEC) and frame interleaving in order to increase robustness
against lost packets.
4.1. Forward Error Correction (FEC)
The simple scheme of repetition of previously sent data is
one way of achieving FEC. Another possible scheme, which is
more bandwidth efficient is to use payload external FEC,e.g.,
RFC2733 [5], which generates extra packets containing repair
data.
The repetition method involves the simple retransmission of
previously transmitted frame-blocks together with the current
frame-block(s). This is done by using a sliding window to
group the speech frame-blocks to send in each payload. Figure
1 illustrates an example.
--+--------+--------+--------+--------+--------+--------+--------+--
| f(n-2) | f(n-1) | f(n) | f(n+1) | f(n+2) | f(n+3) | f(n+4) |
--+--------+--------+--------+--------+--------+--------+--------+--
<---- p(n-1) ---->
<----- p(n) ----->
<---- p(n+1) ---->
<---- p(n+2) ---->
<---- p(n+3) ---->
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<---- p(n+4) ---->
Figure 1: An example of redundant transmission.
In this example each frame-block is retransmitted one time in
the following RTP payload packet. Here, f(n-2)..f(n+4)
denotes a sequence of speech frame-blocks and p(n-1)..p(n+4)
a sequence of payload packets.
The use of this approach does not require signaling at the
session setup. In other words, the speech sender can choose
to use this scheme without consulting the receiver. This is
because a packet containing redundant frames will not look
different from a packet with only new frames. The receiver
may receive multiple copies or versions of a frame for a
certain timestamp if no packet is lost. If multiple versions
of the same speech frame are received, it is RECOMMENDED that
the highest rate be used by the speech decoder.
This redundancy scheme provides the same functionality as the
one described in RFC 2198 "RTP Payload for Redundant Audio
Data" [10]. In most cases the mechanism in this payload
format is more efficient and simpler than requiring both
endpoints to support RFC 2198. If the spread in time required
between the primary and redundant encodings is larger than 5
frame times, the bandwidth overhead of RFC 2198 will be
lower.
The sender is responsible for selecting an appropriate amount
of redundancy based on feedback about the channel, e.g., in
RTCP receiver reports, or network traffic. A sender should
not base selection of FEC on the CMR, as this parameter
most probably was set based on none-IP information. The
sender is also responsible for avoiding congestion, which may
be aggravated by redundant transmission.
4.2. Frame Interleaving and Multi-Frame Encapsulation
To decrease protocol overhead, the octet-aligned payload
format allows several speech frame-blocks to be encapsulated
into a single RTP packet. One of the drawbacks of such
approach is that in case of packet loss this means loss of
several consecutive speech frame-blocks, which usually causes
clearly audible distortion in the reconstructed speech.
Interleaving of frame-blocks can improve the speech quality
in such cases by distributing the consecutive losses into a
series of single frame-block losses. However, interleaving
and bundling several frame-blocks per payload will also
increase end-to-end delay and is therefore not appropriate
for all types of applications. Streaming applications will
most likely be able to exploit interleaving to improve speech
quality in lossy transmission conditions.
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The octet-aligned payload format supports the use of frame
interleaving as an option. For the encoder (speech sender) to
use frame interleaving in its outbound RTP packets for a
given session, the decoder (speech receiver) needs to
indicate its support via out-of-band means (see Section 10).
5. VMR-WB Voice over IP Scenarios
5.1 IP Terminal to IP Terminal
The primary scenario for this payload format is IP end-to-end
between two terminals incorporating VMR-WB codec, as shown in
Figure 2. This payload format is expected to be useful for
both conversational and streaming services.
+----------+ +----------+
| | | |
| TERMINAL |<----------------------->| TERMINAL |
| | VMR-WB/RTP/UDP/IP | |
+----------+ +----------+
Figure 2: IP terminal to IP terminal scenario
A conversational service puts requirements on the payload
format. Low delay is a very important factor, i.e. fewer
speech frame-blocks per payload packet. Low overhead is also
required when the payload format traverses across low
bandwidth links, especially if the frequency of packets will
be high.
Streaming service has less strict real-time requirements and
therefore can use a larger number of frame-blocks per packet
than conversational service. This reduces the overhead from
IP, UDP, and RTP headers. However, including several frame-
blocks per packet makes the transmission more vulnerable to
packet loss, so interleaving may be used to reduce the effect
of packet loss on speech quality. A streaming server handling
a large number of clients also needs a payload format that
requires as few resources as possible when doing
packetization.
Note that all modes of the VMR-WB codec can be used in this
scenario. Also both header-free and octet-aligned payload
formats can be utilized.
5.2 IP Terminal to GW to IP Terminal
A second scenario for this payload format is IP end-to-end
(Through a gateway) between two terminals, one with AMR-WB
codec and the other one with VMR-WB codec using the
interoperable mode of VMR-WB, as shown in Figure 3. This
payload format is expected to be useful for both
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conversational and streaming services.
+----------+ +------+ +----------+
| | VMR-WB/RTP/UDP/IP | | AMR-WB/RTP/UDP/IP | |
| TERMINAL |<-------------------->| GW |<------------------->| TERMINAL |
| | | | | |
+----------+ +------+ +----------+
VMR-WB enabled | AMR-WB enabled
|
|
<----VMR-WB Session----> <----AMR-WB Session---->
Figure 3: IP terminal to GW to IP terminal scenario
(AMR-WB <-> VMR-WB interoperable interconnection)
The VMR-WB mode 3 and octet-aligned payload format SHALL be
used for this scenario. Moreover, to avoid signaling
conflicts in the IP network, two sessions SHALL be
established using SIP/SDP, one between the VMR-WB enabled
terminal and the gateway and another session between the
gateway and the AMR-WB enabled terminal. Note that no
transcoding is involved since the VMR-WB payload is identical
to that of AMR-WB.
5.3 GW to IP Terminal
Another scenario occurs when VMR-WB encoded speech will be
transmitted from a non-IP system (e.g., 3GPP2/CDMA2000
network) to an RTP/UDP/IP VoIP terminal, and/or vice versa,
as depicted in Figure 4.
VMR-WB over
3GPP2/CDMA2000 network
+------+ +----------+
| | | |
<-------------->| GW |<---------------------->| TERMINAL |
| | VMR-WB/RTP/UDP/IP | |
+------+ +----------+
|
| IP network
|
Figure 4: GW to VoIP terminal scenario
VMR-WB's capability to seamlessly switch between operational
modes is exploited in CDMA (non-IP) networks to optimize
speech quality for a given traffic condition. To preserve
this functionality in scenarios including a gateway to an IP
network using the octet-aligned payload format, a codec mode
request (CMR) field is considered. The gateway will be
responsible for forwarding the CMR between the non-IP and IP
parts in both directions. The IP terminal should follow the
CMR forwarded by the gateway to optimize speech quality going
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to the non-IP decoder. The mode control algorithm in the
gateway SHOULD accommodate the delay imposed by the IP
network on the response to CMR by the IP terminal.
The IP terminal should not set the CMR (see Section 6.3.2),
but the gateway can set the CMR value on frames going toward
the encoder in the non-IP part to optimize speech quality
from that encoder to the gateway. The gateway can
alternatively set a different CMR value, if desired, as one
means to control congestion on the IP network.
5.4 GW to GW (Between VMR-WB and AMR-WB Enabled Terminals)
A fourth likely scenario is that RTP/UDP/IP is used as
transport between two non-IP systems, i.e., IP is originated
and terminated in gateways on both sides of the IP transport,
as illustrated in Figure 5. This is the most likely scenario
for an interoperable interconnection between
3GPP/(GSM,WCDMA)/AMR-WB and 3GPP2/CDMA2000/VMR-WB.
VMR-WB over AMR-WB over
3GPP2/CDMA2000 network 3GPP/(GSM, WCDMA) network
+------+ +------+
(VMR-WB Payload) | | AMR-WB/RTP/UDP/IP | | (AMR-WB Payload)
<------------------>| GW |<------------------->| GW |<------------------>
| | | |
+------+ +------+
| |
| IP network |
| |
<---VMR-WB Session----> <---------------AMR-WB Session--------------->
Figure 5: GW to GW scenario (AMR-WB <-> VMR-WB
interoperable interconnection)
The VMR-WB mode 3 and octet-aligned payload format SHALL be
used for this scenario. Moreover, to avoid signaling
conflicts in the IP network, two sessions SHALL be
established using SIP/SDP, one between the VMR-WB enabled
terminal and the gateway and another session between the
gateway and the AMR-WB enabled terminal. Note that no
transcoding is involved since the VMR-WB payload is identical
to that of AMR-WB.
The CMR value may be set in packets received by the gateways
on the IP network side. The gateway should forward to the
non-IP side a CMR value that is the minimum of two values (1)
the CMR value it receives on the IP side; and (2) a CMR value
it may choose for congestion control of transmission on the
IP side.
The details of the traffic control algorithm are left to the
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implementation.
During and upon initiation of an interoperable
interconnection between VMR-WB and AMR-WB, only VMR-WB mode 3
SHALL be used. There are three Frame Types (i.e., FT=0, 1, or
2 see Table 3) within this mode that are compatible with
AMR-WB codec modes 0, 1, and 2, respectively.
If the AMR-WB codec is engaged in an interoperable
interconnection with VMR-WB, the active AMR-WB codec mode set
SHALL be limited to 0, 1, and 2.
5.5 GW to GW (Between two VMR-WB Enabled Terminals)
The fifth example VoIP scenario comprises a RTP/UDP/IP
transport between two non-IP systems, i.e., IP is originated
and terminated in gateways on both sides of the IP transport,
as illustrated in Figure 6. This is the most likely scenario
for Mobile Station-to-Mobile Station (MS-to-MS) Transcoder-
Free (TrFO) interconnection between two 3GPP2/CDMA2000
terminals that both use VMR-WB codec.
VMR-WB over VMR-WB over
3GPP2/CDMA2000 network 3GPP2/CDMA2000 network
+------+ +------+
| | | |
<----------------->| GW |<------------------->| GW |<--------------->
| | VMR-WB/RTP/UDP/IP | |
+------+ +------+
| |
| IP network |
| |
Figure 6: GW to GW scenario (a CDMA2000 MS-to-MS
voice over IP scenario)
6. VMR-WB RTP Payload Formats
For a given session, the payload format can be either header
free or octet-aligned, depending on the mode of operation
that is established for the session via out-of-band means and
the application.
The header-free payload format is designed for maximum
bandwidth efficiency, simplicity, and low latency. Only one
codec data frame can be sent in each header-free payload
format. None of the payload header fields or ToC entries is
present [11].
In the octet-aligned payload format, all the fields in a
payload, including payload header, table of contents entries,
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and speech frames themselves, are individually aligned to
octet boundaries to make implementations efficient.
Note that octet alignment of a field or payload means that
the last octet is padded with zeroes in the least significant
bits to fill the octet. Also note that this padding is
separate from padding indicated by the P bit in the RTP
header.
Between the two payload formats, only the octet-aligned
format has the capability to use the interleaving to make the
speech transport robust to packet loss.
The VMR-WB octet-aligned payload format in the interoperable
mode is identical to that of AMR-WB (i.e., RFC 3267).
Implementations SHOULD support both header-free and octet-
aligned payload formats to increase interoperability.
6.1. RTP Header Usage
The format of the RTP header is specified in [3]. This
payload format uses the fields of the header in a manner
consistent with that specification.
The RTP timestamp corresponds to the sampling instant of the
first sample encoded for the first frame-block in the packet.
The timestamp clock frequency is the same as the sampling
frequency, so the timestamp unit is in samples.
The duration of one speech frame-block is 20 ms for VMR-WB.
For normal wideband operation of VMR-WB, the input/output
sampling frequency is 16 kHz, corresponding to 320 samples
per frame from each channel. Thus, the timestamp is increased
by 320 for VMR-WB for each consecutive frame-block.
For narrowband operation of VMR-WB, the input/output sampling
frequency is 8 kHz, corresponding to 160 encoded speech
samples per frame from each channel. Thus, the timestamp is
increased by 160 for VMR-WB for each consecutive frame-
block while processing narrowband input/output speech
signals. The choice of sampling frequency MUST be indicated
in the beginning of a session (see section 10). The default
input/output sampling rate is 16 kHz. Note that during a
session, the change of sampling rate is not RECOMMENDED.
A packet may contain multiple frame-blocks of encoded speech
or comfort noise parameters. If interleaving is employed, the
frame-blocks encapsulated into a payload are picked according
to the interleaving rules as defined in Section
6.3.2. Otherwise, each packet covers a period of one or more
contiguous 20 ms frame-block intervals. In case the data from
all the channels for a particular frame-block in the period
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is missing, for example at a gateway from some other
transport format, it is possible to indicate that no data is
present for that frame-block rather than breaking a multi-
frame-block packet into two, as explained in Section 6.3.2.
The payload is always made an integral number of octets long
by padding with zero bits if necessary. If additional padding
is required to bring the payload length to a larger multiple
of octets or for some other purpose, then the P bit in the
RTP header MAY be set and padding appended as specified in
[3].
The RTP header marker bit (M) SHALL be always set to 0 if the
VMR-WB codec operates in continuous transmission. When
operating in discontinuous transmission (DTX), the RTP header
marker bit SHALL be set to 1 if the first frame-block carried
in the packet contains a speech frame, which is the first in
a talkspurt. For all other packets the marker bit SHALL be
set to zero (M=0).
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 under
which this payload format is being used will assign a payload
type for this encoding or specify that the payload type is to
be bound dynamically.
6.2. Header-Free Payload Format
The header-free Packet payload format is designed for maximum
bandwidth efficiency, simplicity, and minimum delay. Only one
speech data frame can be sent in each header-free payload
format. None of the payload header fields or ToC entries is
present. The encoding rate for the speech frame can be
determined from the length of the speech data frame, since
there is only one speech data frame in each header-free
payload format.
Use of the RTP header fields for header-free payload format
is the same as the corresponding one for the octet-aligned
payload format. The detailed bit mapping of speech data
packets permissible for this payload format is described in
Section 8 of [1].
Since the header-free payload format is not compatible with
AMR-WB, it is RECOMMENDED that only VMR-WB modes 0, 1, and 2
be used with this payload format.
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] |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
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| |
+ ONLY one speech data frame +-+-+-+-+-+-+-+-+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Note that the mode of operation, using this payload format,
is decided by the transmitting (encoder) site. The default
mode of operation for VMR-WB encoder is mode 0 [1]. The mode
change request MAY also be sent through non-RTP means, which
is out of the scope of this specification.
6.3. Octet-Aligned Payload Format
6.3.1 Payload Structure
The complete payload consists of a payload header, a payload
table of contents, and speech data representing one or more
speech frame-blocks. The following diagram shows the general
payload format layout:
+----------------+-------------------+----------------
| Payload header | Table of contents | Speech data ...
+----------------+-------------------+----------------
6.3.2. The Payload Header
In octet-aligned payload format the payload header consists
of a 4-bit CMR, 4 reserved bits, and optionally, an 8 bit-
interleaving header, as shown below
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+- - - - - - - -
| CMR |R|R|R|R| ILL | ILP |
+-+-+-+-+-+-+-+-+- - - - - - - -
CMR (4 bits): Indicates a codec mode request sent to the
speech encoder at the site of the receiver of this payload,
provided that the network allows the use of the requested
mode.
The value of the CMR field is set according to the following
Table
+-------+------------------------------------------------------------+
| CMR | VMR-WB Operating Modes |
+-------+------------------------------------------------------------+
| 0 | VMR-WB mode 3 (AMR-WB interoperable mode at 6.60 kbps) |
| 1 | VMR-WB mode 3 (AMR-WB interoperable mode at 8.85 kbps) |
| 2 | VMR-WB mode 3 (AMR-WB interoperable mode at 12.65 kbps) |
| 3 | VMR-WB mode 2 |
| 4 | VMR-WB mode 1 |
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| 5 | VMR-WB mode 0 |
| 6-14 | (reserved) |
| 15 | No Preference (Operating mode SHOULD be set by the network)|
+-------+------------------------------------------------------------+
Table 2: List of valid CMR values and their associated VMR-WB
operating modes.
R: is a reserved bit that MUST be set to zero. The receiver
MUST ignore all R bits.
ILL (4 bits, unsigned integer): This is an OPTIONAL field
that is present only if interleaving is signaled out-of-band
for the session. ILL=L indicates to the receiver that the
interleaving length is L+1, in number of frame-blocks.
ILP (4 bits, unsigned integer): This is an OPTIONAL field
that is present only if interleaving is signaled. ILP MUST
take a value between 0 and ILL, inclusive, indicating the
interleaving index for frame-blocks in this payload in the
interleave group. If the value of ILP is found greater than
ILL, the payload SHOULD be discarded.
ILL and ILP fields MUST be present in each packet in a
session if interleaving is signaled for the session.
The mode request received in the CMR field is valid until the
next CMR is received, i.e. a newly received CMR value
overrides the previous one. Therefore, if a terminal
continuously wishes to receive frames in the same mode x, it
needs to set CMR=x for all its outbound payloads, and if a
terminal has no preference in which mode to receive, it
SHOULD set CMR=15 in all its outbound payloads.
If receiving a payload with a CMR value, which is not valid,
the CMR MUST be ignored by the receiver.
In a multi-channel session, CMR SHOULD be interpreted by the
receiver of the payload as the desired encoding mode for all
the channels in the session, if the network allows.
An IP end-point SHOULD NOT set the CMR based on packet losses
or other congestion indications, for several reasons
- The other end of the IP path may be a gateway to a non-IP
network (such as a radio link) that needs to set the CMR
field to optimize performance on that network.
- Congestion on the IP network is managed by the IP sender,
in this case at the other end of the IP path. Feedback
about congestion SHOULD be provided to that IP sender
through RTCP or other means, and then the sender can
choose to avoid congestion using the most appropriate
mechanism. That may include adjusting the codec mode, but
also includes adjusting the level of redundancy or number
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of frames per packet.
The encoder SHOULD follow a received mode request, but MAY
change to a different mode if the network necessitates it,
for example to control congestion.
The CMR field MUST be set to 15 for packets sent to a
multicast group. The encoder in the speech sender SHOULD
ignore mode requests when sending speech to a multicast
session but MAY use RTCP feedback information as a hint that
a mode change is needed.
If interleaving option is utilized, It MUST be performed on a
frame-block basis as oppose to a frame basis in a multi-
channel session.
The following example illustrates the arrangement of speech
frame-blocks in an interleave group during an interleave
session. Here we assume ILL=L for the interleave group that
starts at speech frame-block n. We also assume that the
first payload packet of the interleave group is s and the
number of speech frame-blocks carried in each payload is N.
Then we will have
Payload s (the first packet of this interleave group):
ILL=L, ILP=0,
Carry frame-blocks: n, n+(L+1), n+2*(L+1),..., n+(N-1)*(L+1)
Payload s+1 (the second packet of this interleave group):
ILL=L, ILP=1,
Carry frame-blocks: n+1, n+1+(L+1), n+1+2*(L+1),..., n+1+(N-1)*(L+1)
...
Payload s+L (the last packet of this interleave group):
ILL=L, ILP=L,
Carry frame-blocks: n+L, n+L+(L+1), n+L+2*(L+1), ..., n+L+(N-1)*(L+1)
The next interleave group will start at frame-block n+N*(L+1).
There will be no interleaving effect unless the number of
frame-blocks per packet (N) is at least 2. Moreover, the
number of frame-blocks per payload (N) and the value of ILL
MUST NOT be changed inside an interleave group. In other
words, all payloads in an interleave group MUST have the same
ILL and MUST contain the same number of speech frame-blocks.
The sender of the payload MUST only apply interleaving if the
receiver has signaled its use through out-of-band means.
Since interleaving will increase buffering requirements at
the receiver, the receiver uses MIME parameter
"interleaving=I" to set the maximum number of frame-blocks
allowed in an interleaving group to I.
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When performing interleaving the sender MUST use a proper
number of frame-blocks per payload (N) and ILL so that the
resulting size of an interleave group is less than or equal
to I, i.e., N*(L+1)<=I.
6.3.3. The Payload Table of Contents
The table of contents (ToC) in octet-aligned payload format
consists of a list of ToC entries where each entry
corresponds to a speech frame carried in the payload, i.e.,
+---------------------+
| list of ToC entries |
+---------------------+
When interleaving is used, the frame-blocks in the ToC will
almost never be placed consecutive in time. Instead, the
presence and order of the frame-blocks in a packet will
follow the pattern described in 6.3.2.
The following example shows the ToC of three consecutive
packets, each carrying 3 frame-blocks, in an interleaved two
channel session. Here, the two channels are left (L) and
right (R) with L coming before R, and the interleaving length
is 3 (i.e., ILL=2). This makes the interleave group 9 frame-
blocks large.
Packet #1
---------
ILL=2, ILP=0:
+----+----+----+----+----+----+
| 1L | 1R | 4L | 4R | 7L | 7R |
+----+----+----+----+----+----+
|<------->|<------->|<------->|
Frame- Frame- Frame-
Block 1 Block 4 Block 7
Packet #2
---------
ILL=2, ILP=1:
+----+----+----+----+----+----+
| 2L | 2R | 5L | 5R | 8L | 8R |
+----+----+----+----+----+----+
|<------->|<------->|<------->|
Frame- Frame- Frame-
Block 2 Block 5 Block 8
Packet #3
---------
ILL=2, ILP=2:
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+----+----+----+----+----+----+
| 3L | 3R | 6L | 6R | 9L | 9R |
+----+----+----+----+----+----+
|<------->|<------->|<------->|
Frame- Frame- Frame-
Block 3 Block 6 Block 9
A ToC entry for the octet-aligned payload format is as follows:
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|F| FT |Q|P|P|
+-+-+-+-+-+-+-+-+
The table of contents (ToC) consists of a list of ToC
entries, each representing a speech frame.
F (1 bit): If set to 1, indicates that this frame is followed
by another speech frame in this payload; if set to 0,
indicates that this frame is the last frame in this payload.
FT (4 bits): Frame type index whose value is chosen according
to the following Table.
+----+--------------------------------------------+-------------------+
| FT | Encoding Rate | Frame Size (Bits) |
+----+--------------------------------------------+-------------------+
| 0 | Interoperable Full-Rate (AMR-WB 6.60 kbps) | 132 |
| 1 | Interoperable Full-Rate (AMR-WB 8.85 kbps) | 177 |
| 2 | Interoperable Full-Rate (AMR-WB 12.65 kbps)| 253 |
| 3 | Full-Rate 13.3 kbps | 266 |
| 4 | Half-Rate 6.2 kbps | 124 |
| 5 | Quarter-Rate 2.7 kbps | 54 |
| 6 | Eighth-Rate 1.0 kbps | 20 |
| 7 | (reserved) | |
| 8 | (reserved) | |
| 9 | CNG (AMR-WB SID) | 35 |
| 10 | (reserved) | |
| 11 | (reserved) | |
| 12 | (reserved) | |
| 13 | (reserved) | |
| 14 | Erasure (AMR-WB SPEECH_LOST) | 0 |
| 15 | Blank (AMR-WB NO_DATA) | 0 |
+----+--------------------------------------------+-------------------+
Table 3:VMR-WB payload frame types for real-time
(or non real-time) transport and storage
During the interoperable mode, FT=14 (SPEECH_LOST) and FT=15
(NO_DATA) are used to indicate frames that are either lost or
not being transmitted in this payload, respectively. FT=14 or
15 MAY be used in the non-interoperable modes to indicate
frame erasure or blank frame, respectively (see Section 2.1
of [1]).
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Note that for ToC entries with FT=14 or 15, there will be no
corresponding speech frame in the payload.
Q (1 bit): Frame quality indicator. If set to 0, indicates
the corresponding frame is corrupted. During the
interoperable mode, the receiver side (with AMR-WB codec)
should set the RX_TYPE to either SPEECH_BAD or SID_BAD
depending on the frame type (FT), if Q=0. The VMR-WB encoder
always sets Q bit to 1.
P bits: Padding bits MUST be set to zero.
For multi-channel sessions, the ToC entries of all frames
from a frame-block are placed in the ToC in consecutive.
Therefore, with N channels and K speech frame-blocks in a
packet, there MUST be N*K entries in the ToC, and the first N
entries will be from the first frame-block, the second N
entries will be from the second frame-block, and so on.
6.3.4. Speech Data
Speech data of a payload contains one or more speech as
described in the ToC of the payload.
Each speech frame represents 20 ms of speech encoded in one
of the available encoding rates depending on the operation
mode. The length of the speech frame is defined by the frame
type in the FT field with the following considerations:
- The last octet of each speech frame MUST be padded with
zeroes at the end if not all bits in the octet are used.
In other words, each speech frame MUST be octet-aligned.
- When multiple speech frames are present in the speech
data, the speech frames MUST be arranged one whole frame
after another.
The order and numbering notation of the speech data bits are
as specified in the VMR-WB standard specification [1].
The payload begins with the payload header of one octet or
two if frame interleaving is selected. The payload header is
followed by the table of contents consisting of a list of
one-octet ToC entries.
The speech data follows the table of contents. For
packetization in the normal order, all of the octets
comprising a speech frame are appended to the payload as a
unit. The speech frames are packed in the same order as their
corresponding ToC entries are arranged in the ToC list, with
the exception that if a given frame has a ToC entry with
FT=14 or 15, there will be no data octets present for that
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frame.
6.3.5. Payload Example: Basic Single Channel Payload Carrying Multiple Frames
The following diagram shows an octet-aligned payload format
from a single channel session that carries two VMR-WB Full-
Rate frames (FT=3). In the payload, a codec mode request is
sent (e.g., CMR=4), requesting the encoder at the receiver's
side to use VMR-WB mode 1. No interleaving is 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CMR=4 |R|R|R|R|1|FT#1=3 |Q|P|P|0|FT#2=3 |Q|P|P| f1(0..7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| f1(8..15) | f1(16..23) | ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| r |P|P|P|P|P|P| f2(0..7) | f2(8..15) | f2(16..23) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | l |P|P|P|P|P|P|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
r= f1(264,265)
l= f2(264,265)
Note, in above example the last octet in both speech frames
is padded with zeros to make them octet-aligned.
6.4. Implementation Considerations
An application implementing this payload format MUST
understand all the payload parameters in the out-of-band
signaling used. For example, if an application uses SDP, all
the SDP and MIME parameters in this document MUST be
understood. This requirement ensures that an implementation
always can decide if it is capable or not of communicating.
7. VMR-WB Storage Format
The storage format is used for storing VMR-WB encoded speech
frames in a file or as an e-mail attachment. Multiple channel
content is also supported.
In general, VMR-WB file has the following structure:
+------------------+
| Header |
+------------------+
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| Speech frame 1 |
+------------------+
: ... :
+------------------+
| Speech frame n |
+------------------+
7.1. Single channel Header
A single channel VMR-WB file header contains only a magic
number.
The magic number for single channel VMR-WB files containing
speech data generated in the non-interoperable modes; i.e.,
VMR-WB modes 0, 1, or 2, MUST consist of ASCII character
string
"#!VMR-WB\n"
(or 0x2321564d522d57420a in hexadecimal).
Note, the "\n" is an important part of the magic numbers and
MUST be included in the comparison; otherwise, the single
channel magic number above will become indistinguishable from
that of the multi-channel file defined in the next section.
The magic number for single channel VMR-WB files containing
speech data generated in the interoperable mode; i.e., VMR-WB
mode 3, MUST consist of ASCII character string
"#!VMR-WB_I\n"
(or 0x2321564d522d57425F490a in hexadecimal).
In the interoperable mode, a file generated by VMR-WB is
decodable with AMR-WB (with the exception of different magic
numbers). However, to ensure compatibility and because VMR-WB
can only decode AMR-WB codec modes 0, 1, or 2, AMR-WB codec
SHOULD be instructed not to generate the modes that are not
in common so that files generated by AMR-WB can be decoded by
VMR-WB.
7.2. Multi-channel Header
The multi-channel header consists of a magic number followed
by a 32-bit channel description field, giving the multi-
channel header the following structure:
+----------------------------+
| Magic Number |
+----------------------------+
| Channel Description Field |
+----------------------------+
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The magic number for multi-channel VMR-WB files containing
speech data generated in the non-interoperable modes; i.e.,
VMR-WB modes 0, 1, or 2, MUST consist of the ASCII character
string
"#!VMR-WB_MC1.0\n"
(or 0x2321564d522d57425F4D43312E300a in hexadecimal).
The version number in the magic numbers refers to the version
of the file format.
The magic number for multi-channel VMR-WB files containing
speech data generated in the interoperable mode; i.e., VMR-WB
mode 3, MUST consist of the ASCII character string
"#!VMR-WB_MCI1.0\n"
(or 0x2321564d522d57425F4D4349312E300a in hexadecimal).
The 32-bit channel description field is defined as
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Reserved bits | CHAN |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Reserved bits: MUST be set to 0 when written, and a reader
MUST ignore them.
CHAN (4 bit unsigned integer): Indicates the number of audio
channels contained in this storage file. The valid values and
the order of the channels within a frame-block are specified
in Section 4.1 in [10].
7.3. Speech Frames
After the file header, speech frame-blocks consecutive in
time are stored in the file. Each frame-block contains a
number of octet-aligned speech frames equal to the number of
channels, and stored in increasing order, starting with
channel 1.
Each stored speech frame starts with a one-octet frame header
with the following format:
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|P| FT |Q|P|P|
+-+-+-+-+-+-+-+-+
The FT field is defined as shown in Table 3. The P bits are
padding and MUST be set to 0.
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Q (1 bit): Frame quality indicator. If set to 0, indicates
the corresponding frame is corrupted. The VMR-WB encoder
always sets Q bit to 1.
Following this one octet header, the speech bits are placed
as defined in 6.3.4. The last octet of each frame is padded
with zeroes, if needed, to achieve octet alignment.
The following example shows a VMR-WB speech frame encoded at
Half-Rate (with 124 speech bits) in the storage format.
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| FT=4 |1|0|0| |
+-+-+-+-+-+-+-+-+ +
| |
+ Speech bits for frame-block n, channel k +
| |
+ +
| |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+-+-+-+-+
Frame-blocks or speech frames that are lost in transmission
and thereby not received MUST be stored as Blank/NO_DATA
frames (FT=15) or Erasure/SPEECH_LOST (FT=14) in complete
frame-blocks to keep synchronization with the original media.
8. Congestion Control
The general congestion control considerations for
transporting RTP data apply to VMR-WB speech over RTP as
well. However, the multimode capability of VMR-WB speech
coding may provide an advantage over other payload formats
for controlling congestion since the bandwidth demand can be
adjusted by selecting a different operating mode (i.e., mode
switching).
Another parameter that may impact the bandwidth demand for
VMR-WB is the number of frame-blocks that are encapsulated in
each RTP payload. Packing more frame-blocks in each RTP
payload can reduce the number of packets sent and hence the
overhead from RTP/UDP/IP headers, at the expense of increased
delay.
If forward error correction (FEC) is used to alleviate the
packet loss, the amount of redundancy added by FEC will need
to be regulated so that the use of FEC itself does not cause
a congestion problem.
It is RECOMMENDED that VMR-WB applications using this payload
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format employ congestion control. The actual mechanism for
congestion control is not specified but should be suitable
for real-time transport of datagrams.
9. Security Considerations
RTP packets using the payload formats defined in this
specification are subject to the general security
considerations discussed in [3].
As this format transports encoded speech, the main security
issues include confidentiality and authentication of the
speech itself. The payload format itself does not have any
built-in security mechanisms. External mechanisms, such as
SRTP [8], MAY be used.
This payload format does not exhibit any significant non-
uniformity in the receiver side computational complexity for
packet processing and thus is unlikely to pose a denial-of-
service threat due to the receipt of pathological/corrupted
data.
9.1. Confidentiality
To achieve confidentiality of the encoded VMR-WB speech, all
speech data bits MAY be encrypted. There is no need to
encrypt the payload header or the table of contents due to
the following reasons:
1) They only carry information about the requested speech
mode, frame type, and frame quality
2) This information could be useful to some third party,
e.g., quality monitoring.
As long as the VMR-WB payload is only packed and unpacked at
either end, encryption may be performed after packet
encapsulation so that there is no conflict between the two
operations.
Interleaving may affect encryption. Depending on the
encryption scheme used, there may be restrictions on, for
example, the time when keys can be changed. Specifically, the
key change may need to occur at the boundary between
interleave groups.
The type of encryption method used may impact the error
robustness of the payload data. The error robustness may be
severely reduced when the data is encrypted unless an
encryption method without error-propagation is used, e.g. a
stream cipher.
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9.2. Authentication
To authenticate the sender of the speech, an external
mechanism MUST be used. It is RECOMMENDED that such a
mechanism protect all the speech data bits.
Data tampering by a man-in-the-middle attacker could result
in erroneous depacketization/decoding that could lower the
speech quality. For example, tampering with the CMR field may
result in speech in a different quality than desired.
To prevent a man-in-the-middle attacker from tampering with
the payload packets, some additional information besides the
speech bits SHOULD be protected.
This may include the payload header, ToC, RTP timestamp, RTP
sequence number, and the RTP marker bit.
9.3. Decoding Validation and Provision for Lost or Late Packets
When processing a received payload packet, if the receiver
finds that the calculated payload length, based on the
information of the session and the values found in the
payload header fields, do not match the size of the received
packet, the receiver SHOULD discard the packet to avoid
potential degradation of speech quality and to invoke the
VMR-WB built-in frame error concealment mechanism. Therefore,
invalid packets SHALL be treated as lost packets.
Late packets (i.e., unavailability of a packet when needed
for decoding at the receiver) SHALL be treated as lost
packets. Furthermore, if the late packet is part of an
interleave group, depending upon the availability of the
other packets in that interleave group, decoding MUST be
resumed from the next (sequential order) available packet. In
other words, the unavailability of a packet in an interleave
group at certain time SHOULD not invalidate the other
packets within that interleave group that MAY arrive later.
10. Payload Format Parameters
This section defines the parameters that may be used to
select optional features in the VMR-WB payload. The
parameters are defined here as part of the MIME subtype
registration for the VMR-WB speech codec. A mapping of the
parameters into the Session Description Protocol (SDP) [5] is
also provided for those applications that use SDP. Equivalent
parameters could be defined elsewhere for use with control
protocols that do not use MIME or SDP.
The data format and parameters are specified for both real-
time transport in RTP and for storage type applications such as e-mail
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attachments.
10.1. VMR-WB MIME Registration
The MIME subtype for the Variable-Rate Multimode Wideband
(VMR-WB) audio codec is allocated from the IETF tree since
VMR-WB is expected to be a widely used speech codec in
multimedia streaming and messaging as well as VoIP
applications. This MIME registration covers both real-time
transfer via RTP and non-real-time transfers via stored
files.
Note, the receiver MUST ignore any unspecified parameter and
use the default values instead.
Media Type name: audio
Media subtype name: VMR-WB
Required parameters: none
Note that if no input parameters are defined, the default
values will be used.
Also note that "crc" and "robust-sorting" parameters from RFC
3267 [4] are not applicable to VMR-WB RTP payload and storage
file formats. To ensure compatibility between VMR-WB and
AMR-WB in the interoperable sessions, one SHOULD make sure
that AMR-WB does not utilize crc and robust-sorting (i.e.,
these options are deactivated in the session initiation).
OPTIONAL parameters:
These parameters apply to RTP transfer only.
payload_format: Permissible values are 0 and 1. If 1,
octet-aligned payload format SHALL be used.
If 0 or if not present, header-free payload
format is employed (default).
maxptime: The maximum amount of media, which can be
encapsulated in a payload packet, expressed
as time in milliseconds. The time is
calculated as the sum of the time the media
present in the packet represents. The time
SHALL be an integer multiple of the frame
size. If this parameter is not present, the
sender MAY encapsulate any number of speech
frames into one RTP packet.
interleaving: Indicates that frame-block level
interleaving SHALL be used for the session
and its value defines the maximum number of
frame-blocks allowed in an interleaving
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group (see Section 6.3.1). If this
parameter is not present, interleaving
SHALL not be used. The presence of this
parameter also implies automatically that
octet-aligned operation SHALL be used.
ptime: see RFC2327 [5]. It SHALL be at least one
frame size for VMR-WB.
channels: The number of audio channels. The possible
values and their respective channel order
is specified in section 4.1 in [10]. If
omitted it has the default value of 1.
These parameters apply to both real-time and non-real-time
transfers
dtx: Permissible values are 0 and 1. The default
is 0 (i.e., No DTX) where VMR-WB normally
operates as a continuous variable-rate
codec. If dtx=1, the VMR-WB codec will
operate in discontinuous transmission mode
where silence descriptor (SID) frames are
sent by the VMR-WB encoder during silence
intervals with an adjustable update
frequency. The selection of the SID update-
rate depends on the implementation and
other network considerations that are
beyond the scope of this specification.
Encoding considerations:
This type is defined for transfer via both RTP (RFC
3550) and stored-file methods as described in Sections
6 and 7, respectively, of RFC XXXX. Audio data is
binary data, and must be encoded for non-binary
transport; the Base64 encoding is suitable for Email.
Security considerations:
See Section 9 of RFC XXXX.
Public specification:
The VMR-WB speech codec is specified in following
3GPP2 specifications C.S0052-0 version 1.0.
Transfer methods are specified in RFC XXXX.
Additional information:
The following applies to stored-file transfer methods:
Magic numbers:
Single channel (for the non-interoperable modes)
ASCII character string "#!VMR-WB\n"
(or 0x2321564d522d57420a in hexadecimal)
Single channel (for the interoperable mode)
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ASCII character string "#!VMR-WB_I\n"
(or 0x2321564d522d57425F490a in hexadecimal)
Multi-channel (for the non-interoperable modes)
ASCII character string "#!VMR-WB_MC1.0\n"
(or 0x2321564d522d57425F4D43312E300a in hexadecimal)
Multi-channel (for the interoperable mode)
ASCII character string "#!VMR-WB_MCI1.0\n"
(or 0x2321564d522d57425F4D4349312E300a in hexadecimal)
File extensions for the non-interoperable modes: vmr, VMR
Macintosh file type code: none
Object identifier or OID: none
File extensions for the interoperable mode: vmi, VMI
Macintosh file type code: none
Object identifier or OID: none
Person & email address to contact for further information:
Sassan Ahmadi, Ph.D. Nokia Inc. USA
sassan.ahmadi@nokia.com
Intended usage: COMMON.
It is expected that many VoIP, multimedia messaging and
streaming applications (as well as mobile applications)
will use this type.
Author/Change controller:
Sassan Ahmadi, Ph.D. Nokia Inc. USA
sassan.ahmadi@nokia.com
IETF Audio/Video Transport Working Group
10.2. Mapping MIME Parameters into SDP
The information carried in the MIME media type specification
has a specific mapping to fields in the Session Description
Protocol (SDP) [5], which is commonly used to describe RTP
sessions. When SDP is used to specify sessions employing the
VMR-WB codec, the mapping is as follows:
- The MIME type ("audio") goes in SDP "m=" as the media
name.
- The MIME subtype (payload format name) goes in SDP
"a=rtpmap" as the encoding name. The RTP clock rate in
"a=rtpmap" MUST be 16000 for VMR-WB (Note that 8000 is
also supported by VMR-WB for narrowband I/O processing),
and the encoding parameters (number of channels) MUST
either be explicitly set to N or omitted, implying a
default value of 1. The values of N that are allowed is
specified in Section 4.1 in [10].
- The parameters "ptime" and "maxptime" go in the SDP
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"a=ptime" and "a=maxptime" attributes, respectively.
- Any remaining parameters go in the SDP "a=fmtp" attribute
by copying them directly from the MIME media type string
as a semicolon separated list of parameter=value pairs.
Some example SDP session descriptions utilizing VMR-WB
encodings follow. In these examples, long a=fmtp lines are
folded to meet the column width constraints of this document;
the backslash ("\") at the end of a line and the
carriage return that follows it should be ignored.
Example of usage of VMR-WB in a possible VoIP scenario
(wideband audio):
m=audio 49120 RTP/AVP 98
a=rtpmap:98 VMR-WB/16000
a=fmtp:98 payload_format=1
Example of usage of VMR-WB in a possible VoIP scenario
(narrowband audio):
m=audio 49120 RTP/AVP 98
a=rtpmap:98 VMR-WB/8000
a=fmtp:98
Example of usage of VMR-WB in a possible streaming scenario
(two channel stereo):
m=audio 49120 RTP/AVP 99
a=rtpmap:99 VMR-WB/16000/2
a=fmtp:99 interleaving=30
a=maxptime:100 payload_format=1
10.3. Offer-Answer Model Considerations
To achieve good interoperability for the VMR-WB RTP payload in an
Offer-Answer negotiation usage in SDP the following considerations
SHOULD be made:
- Both header-free and octet-aligned payload formats MAY be offered by
a VMR-WB enabled terminal. However, for an interoperable
interconnection with AMR-WB only octet-aligned payload format SHALL be
used.
- The parameters "maxptime" and "ptime" should in most cases not
affect the interoperability, however the setting of the parameters
can affect the performance of the application.
- To maintain interoperability with AMR-WB in cases where
negotiation is possible using the VMR-WB interoperable mode, a
VMR-WB enabled terminal SHOULD also declare itself capable of AMR-WB
with limited mode set (i.e., only AMR-WB codec modes 0, 1, and
2 are allowed) and octet-align mode of operation. Example:
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m=audio 49120 RTP/AVP 98 99
a=rtpmap:98 VMR-WB/16000/1
a=rtpmap:99 AMR-WB/16000/1
a=fmtp:99 octet-align=1; mode-set=0,1,2
11. IANA Considerations
The new attributes "dtx" and "payload_format" need to be registered.
The definition of the "maxptime" attribute used in this specification is
consistent with the corresponding parameter in RFC 3267.
12. Acknowledgements
The author would like to thank Redwan Salami of VoiceAge
Corporation, Ari Lakaniemi of Nokia Inc., and IETF/AVT chairs Colin
Perkins and Magnus Westerlund for their technical comments
to improve this document.
Also, the author would like to acknowledge that some parts of
RFC 3267 [4] and RFC 3558 [11] have been used in this
document.
References
Normative References
[1] 3GPP2 C.S0052-0 "Source-Controlled Variable-Rate
Multimode Wideband Speech Codec (VMR-WB) Service Option
62 for Wideband Spread Spectrum
Communication Systems", 3GPP2 Technical Specification,
June 2004.
[2] S. Bradner, "Key words for use in RFCs to Indicate
Requirement Levels", IETF RFC 2119, March 1997.
[3] H. Schulzrinne, S. Casner, R. Frederick, and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", IETF RFC 3550, July 2003.
[4] J. Sjoberg, et al., "Real-Time Transport Protocol (RTP)
Payload Format and File Storage Format for the Adaptive
Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
(AMR-WB) Audio Codecs", IETF RFC 3267, June 2002.
[5] M. Handley and V. Jacobson, "SDP: Session Description
Protocol", IETF RFC 2327, April 1998.
Informative References
[6] M. Handley, S. Floyd, J. Padhye, J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
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IETF RFC 3448, January 2003.
[7] J. Rosenberg, and H. Schulzrinne, "An RTP Payload Format
for Generic Forward Error Correction", IETF RFC 2733,
December 1999.
[8] Baugher, et al., "The Secure Real Time Transport
Protocol", IETF Draft (Work in Progress), November 2001.
[9] C. Perkins, et al., "RTP Payload for Redundant Audio
Data", IETF RFC 2198, September 1997.
[10] H. Schulzrinne, "RTP Profile for Audio and Video
Conferences with Minimal Control" IETF RFC 3551, July
2003.
[11] A. Li, "RTP Payload Format for Enhanced Variable Rate
Codecs (EVRC) and Selectable Mode Vocoders (SMV)", IETF
RFC 3558, July 2003.
[12] 3GPP TS 26.193 "AMR Wideband Speech Codec; Source
Controlled Rate operation", version 5.0.0 (2001-03), 3rd
Generation Partnership Project (3GPP).
Any 3GPP2 document can be downloaded from the 3GPP2 web
server, "http://www.3gpp2.org/", see specifications.
Author's Address
The editor will serve as the point of contact for all
technical matters related to this document.
Dr. Sassan Ahmadi Phone: 1 (858) 831-5916
Fax: 1 (858) 831-4174
Nokia Inc. Email: sassan.ahmadi@nokia.com
12278 Scripps Summit Dr.
San Diego, CA 92131 USA
This Internet-Draft expires in six months from May 17, 2004.
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itself may not be modified in any way, such as by removing
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INTERNET-DRAFT VMR-WB RTP Payload & File Storage Formats May 2004
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Sassan Ahmadi [page 32]