Network Working Group G. Zorn, Ed.

Internet-Draft Network Zen

Intended status: Standards Track Y. Wang
Expires: February 17, 2013 Qualcomm Incorporated
A. Lakaniemi
Independent Contributor
August 16, 2012

RTP Payload Format for G.718 Speech/Audio

draft-ietf-payload-rtp-g718-03

Abstract

This document specifies the Real-Time Transport Protocol (RTP)

payload format for the Embedded Variable Bit-Rate (EV-VBR) speech/
audio codec, specified in ITU-T G.718. A media type registration for
this RTP payload format is also included.

Status of This Memo

This Internet-Draft is submitted in full conformance with the

provisions of BCP 78 and BCP 79.

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This Internet-Draft will expire on February 17, 2013.

Copyright Notice

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document authors. All rights reserved.

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the Trust Legal Provisions and are provided without warranty as

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Table of Contents

1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Requirements Language . . . . . . . . . . . . . . . . . . . . 4
3. Background . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3.1. The G.718 Codec . . . . . . . . . . . . . . . . . . . . . 4
3.2. Benefits of Layered Design . . . . . . . . . . . . . . . . 6
3.3. Transmitting Layered Data . . . . . . . . . . . . . . . . 6
3.4. Scaling Scenarios and Rate Control . . . . . . . . . . . . 7
4. G.718 RTP Payload Format . . . . . . . . . . . . . . . . . . . 7
4.1. Payload Structure . . . . . . . . . . . . . . . . . . . . 8
4.1.1. Payload Header . . . . . . . . . . . . . . . . . . . . 8
4.1.2. G.718 Transport Blocks . . . . . . . . . . . . . . . . 8
4.2. Handling The Encoded Data . . . . . . . . . . . . . . . . 11
4.3. G.718 Scaling . . . . . . . . . . . . . . . . . . . . . . 13
4.4. CRC Verification . . . . . . . . . . . . . . . . . . . . . 13
4.5. G.718 Session . . . . . . . . . . . . . . . . . . . . . . 14
4.6. Cross-stream/Cross-layer Timing Synchronization . . . . . 14
4.7. RTP Header Usage . . . . . . . . . . . . . . . . . . . . . 14
5. Payload Format Parameters . . . . . . . . . . . . . . . . . . 15
5.1. Media Type Registration . . . . . . . . . . . . . . . . . 15
5.2. Mapping to SDP Parameters . . . . . . . . . . . . . . . . 17
5.3. Offer/Answer Considerations . . . . . . . . . . . . . . . 17
5.4. Declarative Usage of SDP . . . . . . . . . . . . . . . . . 18
5.5. SDP Examples . . . . . . . . . . . . . . . . . . . . . . . 18
5.5.1. Example 1 . . . . . . . . . . . . . . . . . . . . . . 18
5.5.2. Example 2 . . . . . . . . . . . . . . . . . . . . . . 18
5.5.3. Example 3 . . . . . . . . . . . . . . . . . . . . . . 19
6. Congestion Control . . . . . . . . . . . . . . . . . . . . . . 20
7. Security Considerations . . . . . . . . . . . . . . . . . . . 20
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 21
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 21
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 21
10.1. Normative References . . . . . . . . . . . . . . . . . . . 21
10.2. Informative References . . . . . . . . . . . . . . . . . . 22
Appendix A. Payload Examples . . . . . . . . . . . . . . . . . . 23
A.1. Simple Payload Examples . . . . . . . . . . . . . . . . . 23
A.1.1. All The Layers in The Same Payload . . . . . . . . . . 23
A.1.2. Layers in Seperate RTP Streams . . . . . . . . . . . . 24
A.2. Advanced Examples . . . . . . . . . . . . . . . . . . . . 25
A.2.1. Different Update Rate for Subset of Layers . . . . . . 25
A.2.2. Redundant Frames With Limited Set of Layers . . . . . 26
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1. Introduction

The International Telecommunication Union (ITU-T) Recommendation

G.718 [ITU.G718.2008] specifies the Embedded Variable Bit Rate (EV-
VBR) speech/audio codec. This document specifies the Real-time
Transport Protocol (RTP) [RFC3550] payload format for this codec.

2. Requirements Language

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].

3. Background

3.1. The G.718 Codec

G.718 is an embedded variable rate speech codec having a layered

design. The bitstream of the G.718 core codec consists of a core
layer, denoted as L1, and four enhancement layers, denoted as L2-L5.
The bit-rates of the G.718 core codec range from 8 kbit/s (core layer
only) to 32 kbit/s (with all layers up to L5). Furthermore, the
G.718 codec also supports discontinuous transmission (DTX) and
comfort noise generation (CNG) by sending Silence Descriptor (SID)
frames during periods of non-active input signal, resulting in a
reduced bit-rate. The sampling frequency of the core codec is 16 kHz
and the codec operates on 20 ms frames. The G.718 codec is also
capable of narrowband operation with audio input and/or output at 8
kHz sampling frequency.

While transmitting/receiving the core layer L1 is enough for

successful decoding of the audio content, each of the enhancement
layers Ln (n being 2 to 5, inclusive) provides an improvement to
reconstructed audio quality. Thus, the core layer ensures the basic
communication while the enhancement layers can be used to improve the
perceptual quality. Furthermore, enhancement layers are dependent on
all the lower layers in a sense that successful decoding of layer Ln
requires also all the layers Lm with m<n to be available.

The sizes, sampling rates and possible outputs of the G.718 core
codec layers L1-L5 are summarized in Table 1 below, where the "Bytes"
column indicates the number of bytes per encoded data unit for a
layer. NB and WB denote narrowband and wideband, respectively. The
"Bytes" column in other tables has the same meaning. Note that for
layers L1 and L2, the corresponding output may either be NB or WB,
depending on the rendering device and the application requirement,
regardless of the sampling rate of the encoded data.
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Table 1: G.718 Layers

Layer Bytes Cumulative bit-rate Sampling rate Output

----------------------------------------------------------------
L1' 32 12.8 kbit/s 16 kHz WB
L3' 9 16.4 kbit/s 16 kHz WB
L4 20 24.4 kbit/s 16 kHz WB
L5 20 32.4 kbit/s 16 kHz WB

The G.718 codec also includes an operating mode that is compatible

with the Adaptive Multi-Rate Wideband (AMR-WB) codec [AMR-WB], for
which the RTP payload format is specified in [RFC4867]. In this AMR-
WB interoperable mode, layers L1 and L2 are replaced by L1'
consisting of AMR-WB encoded data and L3' is used instead of L3. The
usage of layers L4 and L5 is not affected by transmitting AMR-WB data
in the lower layers. If layer L3' is present in the encoded bit-
stream, the base layer L1' must use the AMR-WB mode 2 with a bit-rate
of 12.65 kbits/s. Otherwise (the encoded bit-stream contains only
the L1' layer), any of the 9 AMR-WB coding modes 0, 1, 2, 3, 4, 5, 6,
7, and 8 correspond to the bit- rates of 6.60, 8.85, 12.65, 14.25,
15.85, 18.25, 19.85, 23.05, and 23.85 kbit/s, respectively, may be in
use. Table 2 summarizes the AMR-WB interoperable mode when more than
one layer may be present.

Table 2: G.718 layers in the AMR-WB interoperable mode

Layer Bytes Cumulative bit-rate Sampling rate Output

----------------------------------------------------------------
L1' 32 12.8 kbit/s 16 kHz WB
L3' 9 16.4 kbit/s 16 kHz WB
L4 20 24.4 kbit/s 16 kHz WB
L5 20 32.4 kbit/s 16 kHz WB

Note that the bit-rate for the raw bit-stream of AMR-WB mode 2 is
12.65 kbits/s. However, after counting the padding bits to make each
encoded data unit byte-aligned, as in the octet-aligned mode
specified in [RFC4867], the resulting bit-rate is then 12.8 kbits/s.

In the AMR-WB interoperable mode, when the base layer L1' is

transported in its own RTP packet stream, the packetisation specified
in [RFC4867] MUST be used, to enable legacy RFC4867 receivers to
receive the base layer L1'.

ITU-T SG16 is currently working on a set of extension layers in order

to provide so-called super-wideband (SWB) audio and stereophonic
encoding extensions on top of the G.718 core codec. Further details
and the usage of these layers are undtermined at this time.
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The main application of the G.718 codec is telephony. Other expected

applications include audio/video conferencing and streaming.

3.2. Benefits of Layered Design

Layered design enables simple scalability of the transmitted stream

simply by conveying a suitable number of layers. The number of
layers used in a session may be selected for example based on the
capacity of the transmission channel, current transmission
conditions, characteristics of the source signal or available
processing capacity.

Another obvious benefit of the layered codec design is the

possibility to exploit the scalability to support congestion control
by transmitting/dropping some of the (higher) enhancement layers in
order to alleviate congestion in the network. See more detailed
discussion on the congestion control in Section 6.

Furthermore, the layered design also implicitly provides possibility

for unequal error detection/protection by employing different levels
of protection on core layer and enhancement layers.

3.3. Transmitting Layered Data

In principle there are two basic approaches to carry the data from a
layered encoder:

1. All the layers are carried within a single RTP session

2. The encoded data is divided over multiple RTP sessions, each

session carrying a subset of layers. This is also referred to as
Multi-Session Transmission (MST)

The first choice is the most efficient in terms of exploitation of

transmission bandwidth. Furthermore, using only one packet to carry
all encoded data layers of a frame requires less resources also from
the end-systems (and intermediate systems) since the number of
packets is kept at minimum and only single RTP packet stream needs to
be handled. However, this option requires any intermediate network
element performing the scaling operation to be fully media-aware
since removing encoded layers requires modification of the payload.
Furthermore, the intermediate network element needs to be within the
security context to enable the meaningful manipulation of the
payload, in case secure transport is employed. This might not be
feasible in all systems/scenarios, but some special-purpose devices
such as e.g. media gateways in cellular telephone systems may be able
to implement this kind of media-aware functionality.
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The second alternative, transmitting selected subsets of layers in

separate RTP sessions, facilitates simple scalability in intermediate
network elements without the requirement of being fully media-aware.
One use case of this alternative is layered multicast [McCanne]. On
the other hand, this approach introduces separate packet header
overhead for each subset of layers for those low-delay application
scenarios wherein aggregation of data from multiple frames is not
ideal. In this case, when the size of the encoded data block per
single layer is in the range of 10 to 20 bytes, the packetisation may
result in relatively high amount of protocol overhead, which might be
an expensive solution on bandwidth-limited links. Another drawback
of this approach is somewhat more complex session setup and the
additional complexity associated with handling of several concurrent
RTP sessions. However, this is a trade-off that enables simple
scalability also by intermediate network elements that are not aware
of the details of the transmitted media.

3.4. Scaling Scenarios and Rate Control

In principle there are three different ways to make use of the

layered design to control the bandwidth usage:

1. A sender decides to change the number of layers it is

transmitting (for example due to congestion control constraints)

2. A receiver or an intermediate network element instructs a sender

to change the number of layers it is transmitting

3. An intermediate network element passes through only a subset of

layers it receives

The most appropriate mechanism depends on the application and the

employed network topology. For example point-to-point conversational
audio connection can easily introduce rate control by changing the
number of transmitted layers, while in centralized audio/video
conferencing scenario the conference server is a more appropriate
point to implement the rate control instead of transmitting end-
point. Please refer to RFC 5117 for extensive discussion on the
different topologies and their implications to the transmission.
However, the fundamental difference between these choices is that
method 1 does not necessarily need any feedback from the receiver(s),
while methods 2 and 3 require a signaling mechanism to support rate
control.

4. G.718 RTP Payload Format

The basic G.718 source data unit is one layer of an encoded frame.
Since generally the term layer refers to time series of data
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representing certain encoding layer, in this specification we use the

term Encoded Data Unit (EDU) to refer to a single layer of data from
single encoded frame. Thus, each EDU has a (conceptual) frame number
indicating its location in encoding/decoding order and a layer number
indicating the encoding layer the EDU represents.

4.1. Payload Structure

The G.718 payload format consists of a payload header, followed by

one or more transport blocks (TB) forming the actual payload data.

+-----------------+----------+----------+- /// -+----------+

| Payload header | TB(1) | TB(2) | TB(n) |
+-----------------+----------+----------+- /// -+----------+

4.1.1. Payload Header

The payload header consists of an 8-bit payload CRC checksum:

+-+-+-+-+-+-+-+-+
| CRC |
+-+-+-+-+-+-+-+-+

On the transmitting end the payload checksum is computed over the

primary transport block (specified in Section 4.1.2) of the payload
using the generator polynomial

C(z) = z^8 + z^4 + z^3 + z^2 + 1

Subsequent transport blocks are prepared in such a way that the

payload checksum is valid for any integer number of contiguous
transport blocks within one RTP packet starting from the beginning of
the primary transport block.

On the receiving end the payload CRC checksum can be used to verify
the correct reception of any contiguous subset of transport blocks
within one RTP packet starting from the beginning of the primary
transport block (see Section 4.4 for a detailed description).

4.1.2. G.718 Transport Blocks

The basic building block of the G.718 RTP payload data is an G.718
transport block (TB). There are two types of transport blocks:
primary and secondary.
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The structure of the primary transport block is depicted below.

0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+----------------------------+
| L-ID |NF | Encoded data |
+-+-+-+-+-+-+-+-+----------------------------+

The structure of the secondary transport block is depicted below.

0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+----------------------------+-+-+-+-+-+-+-+-+
| L-ID |NF | Encoded Data | Tail |
+-+-+-+-+-+-+-+-+----------------------------+-+-+-+-+-+-+-+-+

The layer ID (L-ID) and the NF fields form the transport block
header. The L-ID field is used to identify the layer structure of
the encoded data carried in this G.718 transport block, and the NF
field indicates the number of encoded frames with this layer
structure carried in the Encoded data part following the transport
block header. The Tail field of the secondary transport block
carries a modified 8-bit CRC checksum computed over the transport
block, as specified below.

A G.718 RTP packet payload SHALL include exactly one primary

transport block, which MAY be followed by one or more secondary
transport blocks. The data fields of both transport block types are
described below.

L-ID (6 bits)
Identification of the encoded data carried in this transport
block. Table 3 below specifies the mapping between L-ID and the
encoded data. Note that L-ID is treated as an unsigned integer.
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Table 3: Layer Identification (L-ID) Values

L-ID Encoded data

--------------------------
0 Empty frame
1 L1
2 L1-L2
3 L1-L3
4 L1-L4
5 L1-L5
6 L2
7 L2-L3
8 L2-L4
9 L2-L5
10 L3
11 L3-L4
12 L3-L5
13 L4
14 L4-L5
15 L5
16 L1'
17 L1', L3'
18 L1', L3', L4
19 L1', L3', L4-L5
20 G.718 SID
21 AMR-WB SID
22-63 Reserved

NF (2 bits)
Number of frames in this transport block (2 bits) decreased by
one. The number of frames is equal to the value of NF incremented
by one. For example, value NF=0 indicates that the transport
block carries one frame, and value NF=3 indicate that the
transport block carries four frames. If the sender wants to
encapsulate more than four frames per payload, several transport
blocks need to be used.

Encoded Data (variable length)

Encoded data consists of EDUs as specified by the values L-ID and
NF fields, arranged according to the rules given in Section 4.2.
When L-ID is equal to 0 (empty frame), the encoded data field is
not present.

Tail (8 bits)
The Tail field of the secondary transport block carries a bit
field that is needed to modify the partial CRC checksum over the
payload data up to the end of this TB to match the payload CRC
field value carried in the payload header.
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In the transmitter the Tail bits for a secondary TB(n) are

computed by first computing the CRC checksum CRC(n) over the
payload data from the beginning of the primary TB up to the end of
TB(n) using the generator polynomial C(z) given above. The bits
of the Tail field of TB(n) are set to zero value for the CRC
computation. The transmitted value of the Tail field in TB(n) is
obtained by bitwise XOR operation between the payload CRC field
value carried in the payload header and the CRC(n) computed for
TB(n).

4.2. Handling The Encoded Data

In order to provide unique mapping of EDUs to encoded frames, the

following rules on sequence of frames and sequence of layers need to
be followed when creating a payload:

o The frames within a payload MUST form a set of contiguous frames

in decoding order, i.e. if a payload carries frames n and n+N, all
frames between n and n+N in decoding order MUST also be present in
the payload.

o The layers within a frame MUST form a contiguous set of layers,

i.e. if layers Lx and Ly of a frame are included in the payload,
all layers between Lx and Ly layers MUST also be present.

The EDUs within a transport block are arranged according to the

following rules:

o The EDUs within a transport block MUST be arranged in increasing

order of layer number

o The EDUs with the same layer number within a transport block MUST
be arranged in decoding order

Explicit timing information for the transport blocks is not needed,

since the ordering of EDUs in the payload and their mapping to
transport blocks can be used to implicitly carry this information.
The following rules apply:

o If the highest layer carried in transport block k is n, and the

lowest layer carried by transport block k+1 is n+1, then the EDUs
of transport block k and k+1 belong to the same encoded frame.
Furthermore, if transport blocks k and k+1 carry EDUs belonging to
the same encoded frame(s), these transport blocks MUST include the
same number of EDUs
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o If the highest layer carried in transport block k is n, and the

lowest layer carried by transport block k+1 is smaller than or
equal to n, the EDUs of transport block k and k+1 belong to the
two separate encoded frames, which are contiguous in decoding
order

o Multiple copies of an EDU MUST NOT be included in the payload

A set of EDUs can be allocated to transport blocks in several ways.

For example each EDU can be encapsulated in its own transport block,
all EDUs can be carried in single transport block, EDUs belonging to
the same encoded frame can be encapsulated in dedicated transport
block, or EDUs representing the same layer can be carried in their
own transport blocks. Three examples on this with two frames with
layers L1-L3 are given below. The first example illustrates the case
using a single transport block for the whole payload, while the
second payload example introduces separate transport blocks for each
of the EDUs. The third example shows an approach where all layers
are carried in dedicated transport blocks. The notation Fx-Ly is
used to denote layer y of frame x.

Example 1: All EDUs in a single transport block

+---------+-----+-------+-------+-------+-------+-------+--------+
| L-ID=3 |NF=1 | F1-L1 | F2-L1 | F1-L2 | F2-L2 | F1-L3 | F2-L3 |
+---------+-----+-------+-------+-------+-------+-------+--------+

Example 2: All EDUs in separate transport blocks

+---------+-----+-------+---------+-----+-------+
| L-ID=1 |NF=0 | F1-L1 | L-ID=1 |NF=0 | F2-L1 |
+---------+-----+-------+---------+-----+-------+
| L-ID=8 |NF=0 | F1-L2 | L-ID=8 |NF=0 | F2-L2 |
+---------+-----+-------+---------+-----+-------+
| L-ID=14 |NF=0 | F1-L3 | L-ID=14 |NF=0 | F2-L3 |
+---------+-----+-------+---------+-----+-------+

Example 3: Dedicated transport for EDUs of each layer

+---------+-----+-------+-------+---------+-----+-------+-------+
| L-ID=1 |NF=1 | F1-L1 | F2-L1 | L-ID=6 |NF=1 | F1-L2 | F2-L2 |
+---------+-----+-------+-------+---------+-----+-------+-------+
| L-ID=10 |NF=1 | F1-L3 | F2-L3 |
+---------+-----+-------+-------+

While the first example carrying data from all layers in the same
transport block obviously consumes less bandwidth, the second example
using separate transport block for each EDU, and the third example
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using dedicated transport blocks for each layer provide simple

scaling possibility: while in the first case the removal of e.g.
layer L3 (from each frame in the payload) would require changing the
value of the L-ID in addition to removing the corresponding EDU(s),
in the second and third options it is enough to just remove all
transport blocks carrying L3 data and the remaining part of the
payload can be left untouched (however the packet size information in
high-layer protocol headers needs change).

4.3. G.718 Scaling

Some Media-Aware Network Elements (MANEs) MAY modify the G.718

bitstream by dropping some of the layers in case congestion control
or e.g. access link bandwidth requires such scaling to take place.
Such MANEs are RTP translators (with the topology Topo-Translator as
described in [RFC5117], for which the rules for RTP translators
specified in [RFC3550] apply.

A payload can be either completely dropped or some of the transport

blocks it carries can be discarded. In case full payloads are
dropped to implement scaling, a packet containing the core layer L1
SHOULD NOT be discarded, since the decoding of higher layers of the
same encoded frame is not possible without the core layer data being
available. This means that payloads with L-ID values equal to 1 to
5, inclusive and 16 to 19, inclusive, SHOULD NOT be completely
discarded.

Author's note: To be checked whether the case of dropping a subset

of the transport blocks in one packet also strictly follows the
topology Topo-Translator.

In case the payload is forwarded with modified content, at least the

primary transport block MUST be preserved in the payload, while some
of the secondary transport blocks at the end of the payload MAY be
discarded.

4.4. CRC Verification

Both UDP-Lite [RFC3828] and DCCP [RFC4340] provide partial checksum

options, in which partially damaged payloads can be delivered to the
application layer. In cases wherein such a transport layer operation
is in use, and the partial checksum service by the transport layer
protects up to the RTP header and the payload header, the CRC
checksum provided in the payload header can be used to verify whether
an RTP packet payload contains corrupt transport blocks.

On the receiving end the CRC verification is made in such a way that
the CRC computation is started from the beginning of the primary TB,
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i.e. from the MSB of the first octet of the TB(1), and the
computation is continued until the end of the payload data or until
an erroneous TB is encountered. At the end of each TB a check MAY be
performed: if the CRC value at the end of TB(n) matches the payload
CRC value received in the payload header, the verification is
successful and the data up TB(n) is valid. If the CRC value at the
end of TB(n) does not match the payload CRC value received in the
payload header, there is an error in the TB(n) and it MUST be
discarded as corrupted. Furthermore, if the verification indicates
corrupted TB(n), all subsequent transport blocks TB(m) with m>n MUST
also be discarded.

4.5. G.718 Session

A G.718 session consists of one or several RTP sessions carrying

G.718 data encoded according to the payload format specified in
Section 4.1.

4.6. Cross-stream/Cross-layer Timing Synchronization

In the case where a G.718 session consists of multiple RTP sessions,

the RTP packets transmitted on separate RTP sessions need to be
synchronized in order to enable reconstruction of the frames in the
receiving end [RFC6051]. Since each of the RTP sessions uses its own
random initial value for the RTP timestamp, there is also a random
offset between the RTP timestamps values carrying the EDUs belonging
to the same encoded frame in different RTP sessions.

The receiver MUST use the traditional RTCP-based mechanism to

synchronize streams by using the RTP and NTP timestamps of the RTCP
Sender Reports (SR) it receives [RFC3550].

4.7. RTP Header Usage

This section specifies the usage of some fields of the RTP header
(specified in Section 5 of [RFC3550]) with the G.718 RTP payload
format. The settings for other RTP header fields are as specified in
[RFC3550].

The RTP timestamp corresponds to the sampling instant of the first

encoded sample of the earliest frame in the payload. The timestamp
clock frequency is 32 kHz.

The marker bit (M) of each of the RTP streams of the session SHALL be
set to value 1 if the payload carries an EDU belonging to the first
frame after an inactive period, i.e. an EDU from the first frame of a
talkspurt. For all other packets the marker bit is set to value 0.
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5. Payload Format Parameters

This section defines the parameters that may be used to configure

optional features in the G.718 RTP transmission.

The parameters are defined here as part of the media subtype

registration for the G.718 codec. Mapping of the parameters into the
Session Description Protocol (SDP) [RFC4566] is also provided for
those applications that use SDP. In control protocols that do not
use MIME or SDP, the media type parameters MUST be mapped to the
format used with that control protocol.

5.1. Media Type Registration

This registration is done using the template defined in RFC 4288

[RFC4288] and following RFC 4855 [RFC4855].

Type name: audio

Subtype name: G718

Required parameters: none

Optional parameters:

mode: This parameter MAY be used to indicate whether the

mode with layer L1 being present or the AMR-WB
compatible mode (with layer L1' being present) is
in use. If this parameter is not present or the
value of this parameter is equal to 0, the mode
with layer L1 being present is in use. Otherwise,
the AMR-WB compatible mode is in use. When this
parameter is present, the value MUST be either 0
or 1.
NOTE: When the upcoming stereo and SWB options are
present, the semantics of this parameter may
change.

layers: The numbers of the layers (in range from 1 to 5,

denoting layers from L1 to L5, respectively)
transmitted in this session, expressed as comma-
separated list of layer numbers. If the parameter
is present, at least layer L1 or L1' MUST be
included in the list of layers in one of the RTP
sessions included in the G.718 session. If the
parameter is not present, all layers up to layer
L5 MAY be used in the session.
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NOTE: Why not use semantics similarly as L-ID?

ptime: The recommended length of time (in milliseconds)

represented by the media in a packet. See Section
6 of [RFC4566].

maxptime: The maximum length of time (in milliseconds) that

can be encapsulated in a packet. See Section 6 of
[RFC4566].

NOTE: Some further study is needed to see if separate parameters

for sending and receiving capabilities/preferences are
needed -- especially for upcoming stereo and SWB options.

NOTE: Support for upcoming SWB and stereo options needs to be

taken into account. Basically we can either 1) extend the
parameter "layers" to cover also this aspect, or 2) define
separate parameter(s) for these new options when more
details on the stereo/SWB support are available.

Encoding considerations:
This media type is framed and contains binary data; see Section
4.8 of [RFC4288].

Security considerations: See Section 7 of RFC XXXX.

[RFC Editor: Upon publication as an RFC, please "XXXX" with the
number assigned to this document and remove this note.]

Interoperability considerations: None.

Published specification: RFC XXX.

[RFC Editor: Upon publication as an RFC, please "XXXX" with the
number assigned to this document and remove this note.]

Applications which use this media type:

For example: Voice over IP, audio and video conferencing, audio
streaming and voice messaging.

Additional information: None.

Person & email address to contact for further information:

Ari Lakaniemi, [email protected]

Intended usage: COMMON

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Restrictions on usage:
This media type depends on RTP framing, and hence is only defined
for transfer via RTP [RFC3550].

Author: Ari Lakaniemi, [email protected]

Change controller:
IETF Audio/Video Transport Working Group delegated from the IESG.

5.2. Mapping to SDP Parameters

The information carried in the media type specification has a

specific mapping to fields of the SDP [RFC4566], which is commonly
used to describe RTP sessions. When SDP is used to specify sessions
employing the G.718 codec, the mapping is as follows:

o The media type ("audio") goes in SDP "m=" as the media name.

o The media subtype ("G718") goes in SDP "a=rtpmap" as the encoding

name. The RTP clock rate in "a=rtpmap" MUST be 32000 for G.718.
NOTE: The current choice for the RTP clock rate is a
'placeholder'. The clock rate needs to be set according to SWB
sampling rate, which is still T.B.D. Since the core codec
employs 16000 Hz sampling rate, an integer multiple of 16000 Hz
seems to be a preferable choice.

o The parameters "ptime" and "maxptime" go in the SDP "a=ptime" and

"a=maxptime" attributes, respectively.

o Any remaining parameters go in the SDP "a=fmtp" attribute by

copying them directly from the media type string as a semicolon
separated list of parameter=value pairs.

5.3. Offer/Answer Considerations

The following considerations apply when using the SDP offer/answer

[RFC3264] mechanism to negotiate the G.718 transport. The parameter
"layers" MAY be used to indicate the layer configuration for the each
RTP session belonging to current G.718 session an end-point making
the offer is ready to transmit and wishes to receive.

o In case the G.718 session consists of a single RTP session, it is

RECOMMENDED not to impose any layer restrictions for the session
but to use the rate control functionality to set possible
restrictions on usage of the higher or highest layers. If the
offer includes a layer configuration parameter, the answer MAY use
different configuration, but the highest layer in the answer MUST
NOT be higher than the highest layer of the offered configuration.
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NOTE: Support for answer modifying the layer configuration is

FFS.

o In case the G.718 session consists of multiple RTP sessions, the

answer MUST use the layer configurations provided in the offer for
the sessions it accepts.

5.4. Declarative Usage of SDP

In declarative usage, such as SDP in RTSP [RFC2326] or SAP [RFC2974],

the parameter "layers" SHALL be interpreted to provide a set of
layers that the sender MAY use in the session.

5.5. SDP Examples

Some example SDP session descriptions utilizing G.718 encodings are

provided below.

5.5.1. Example 1

The first example illustrates the simple case with the G.718 session
employing a single RTP session and the AVPF profile is offered, and
the answer accepts the offer without any changes.

Offer:

m=audio 49120 RTP/AVPF 97

a=rtpmap:97 G718/32000/1

Answer:

m=audio 49120 RTP/AVPF 97

a=rtpmap:97 G718/32000/1

5.5.2. Example 2

This example shows a bit more complex case where the G.718 session
using a single RTP session and the AVPF profile is offered with the
restriction to send/receive only with layers L1 and L2. The answer
indicates that the other end-point is happy to receive (and send)
layers up to L5.

Offer:

m=audio 49120 RTP/AVPF 97

a=rtpmap:97 G718/32000/1
a=fmtp:97 layers=1,2
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Answer:

m=audio 49120 RTP/AVPF 97

a=rtpmap:97 G718/32000/1
a=fmtp:97 layers=1,2,3,4,5

5.5.3. Example 3

The third example shows an G.718 session using multiple RTP sessions
with the AVPF profile. The answerer wishes to use only layers up to
L3.

Offer:

m=audio 49120 RTP/AVPF 97

a=rtpmap:97 G718/32000/1
a=fmtp:97 layers=1,2
a=mid=1

m=audio 49122 RTP/AVPF 98

a=rtpmap:98 G718/32000/1
a=fmtp:98 layers=3
a=mid=2
a=depend:lay 1

m=audio 49124 RTP/AVPF 99

a=rtpmap:99 G718/32000/1
a=fmtp:99 layers=4,5
a=mid=3
a=depend:lay 1 2

Answer:

m=audio 49120 RTP/AVPF 97

a=rtpmap:97 G718/32000/1
a=fmtp:97 layers=1,2
a=mid=1

m=audio 49120 RTP/AVPF 98

a=rtpmap:98 G718/32000/1
a=fmtp:98 layers=3
a=mid=2
a=depend:lay 1

Note that the dependency signaling described in [RFC5583] is used in

the third example above to indicate the relationship between the
layers distributed into separate RTP sessions.
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6. Congestion Control

As a scalable codec, G.718 implicitly provides means for congestion

control by providing a possibility for 'thinning' the bitstream. The
RTP payload format according to this specification provides several
different means for reducing the G.718 session bandwidth. The most
appropriate mechanism (in terms of impact to the user experience)
depends on the employed payload structure and also on the employed
session configuration (single RTP session or multiple RTP sessions).
The following means (in no particular order) can be used to assist
congestion control procedures -- either by the sender or by the
intermediate node.

o The payloads carrying the EDUs representing the highest layers in

an G.718 session can be dropped, along with all associated
transport blocks

o The transport blocks carrying the EDUs representing the highest

layers within the payload can be dropped

o Transport blocks or payloads carrying EDUs belonging to redundant

frames included in the payload can be dropped

7. Security Considerations

RTP packets using the payload format defined in this specification

are subject to the security considerations discussed in the RTP
specification [RFC3550], and in any appropriate RTP profile (for
example [RFC3551] or [RFC4585]. This implies that confidentiality of
the media streams is achieved by encryption; for example, through the
application of SRTP [RFC3711]. Because the data compression used
with this payload format is applied end-to-end, any encryption needs
to be performed after compression.

A potential denial-of-service threat exists for data encodings using

compression techniques that have non-uniform receiver-end
computational load. The attacker can inject pathological datagrams
into the stream that will increase the processing load of the decoder
and may cause the receiver to be overloaded. For example inserting
additional EDUs representing the higher enhancement layers on top of
the ones actually transmitted may increase the decoder load.
However, the G.718 codec is not particularly vulnerable to such an
attack, since the majority of the computational load in an G.718
session is associated to the encoder. Another form of possible
attach might be forging of codec bit-rate control messages, which may
result in encoder operating employing higher number of enhancement
layers than originally intended and thereby requiring larger amount
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of computation resources. Therefore, the usage of data origin

authentication and data integrity protection of at least the RTP
packet is RECOMMENDED; for example, with SRTP [RFC3711].

Note that the appropriate mechanism to ensure confidentiality and

integrity of RTP packets and their payloads is very dependent on the
application and on the transport and signaling protocols employed.
Thus, although SRTP is given as an example above, other possible
choices exist.

Note that end-to-end security with either authentication, integrity

or confidentiality protection will prevent a network element not
within the security context from performing media-aware operations
other than discarding complete packets. To allow any (media-aware)
intermediate network element to perform its operations, it is
required to be a trusted entity which is included in the security
context establishment.

8. IANA Considerations

IANA is kindly requested to register a media type for the G.718 codec
for RTP transport, as specified in Section 5.1 of this document.

9. Acknowledgements

Thanks to Qin Wu for useful review and commentary.

10. References

10.1. Normative References

[AMR-WB] 3GPP, "Speech codec speech processing functions;

Adaptive Multi-Rate - Wideband (AMR-WB) speech
codec; General description", 3GPP TS 26.171 5.0.0,
April 2001.

[ITU.G718.2008] International Telecommunications Union, "Frame Error

Robust Narrowband and Wideband Embedded Variable
Bit-Rate Coding of Speech and Audio from 8-32
Kbit/s", ITU-T Recommendation G.718, May 2008.

[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate

Requirement Levels", BCP 14, RFC 2119, March 1997.

[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer

Model with Session Description Protocol (SDP)",
RFC 3264, June 2002.
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[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.

Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.

[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for

Audio and Video Conferences with Minimal Control",
STD 65, RFC 3551, July 2003.

[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E.,

and K. Norrman, "The Secure Real-time Transport
Protocol (SRTP)", RFC 3711, March 2004.

[RFC4288] Freed, N. and J. Klensin, "Media Type Specifications

and Registration Procedures", BCP 13, RFC 4288,
December 2005.

[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP:

Session Description Protocol", RFC 4566, July 2006.

[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and
J. Rey, "Extended RTP Profile for Real-time
Transport Control Protocol (RTCP)-Based Feedback
(RTP/AVPF)", RFC 4585, July 2006.

[RFC4855] Casner, S., "Media Type Registration of RTP Payload

Formats", RFC 4855, February 2007.

[RFC4867] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q.

Xie, "RTP Payload Format and File Storage Format for
the Adaptive Multi-Rate (AMR) and Adaptive Multi-
Rate Wideband (AMR-WB) Audio Codecs", RFC 4867,
April 2007.

[RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding

Dependency in the Session Description Protocol
(SDP)", RFC 5583, July 2009.

10.2. Informative References

[McCanne] McCanne, S., Jacobson, V., and M. Vetterli,

"Receiver-driven layered multicast", ACM SIGCOMM
Computer Communication Review Volume 26 Issue 4,
October 1996.

[RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real

Time Streaming Protocol (RTSP)", RFC 2326,
April 1998.
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[RFC2974] Handley, M., Perkins, C., and E. Whelan, "Session

Announcement Protocol", RFC 2974, October 2000.

[RFC3828] Larzon, L-A., Degermark, M., Pink, S., Jonsson,

L-E., and G. Fairhurst, "The Lightweight User
Datagram Protocol (UDP-Lite)", RFC 3828, July 2004.

[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram

Congestion Control Protocol (DCCP)", RFC 4340,
March 2006.

[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies",

RFC 5117, January 2008.

[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation

of RTP Flows", RFC 6051, November 2010.

Appendix A. Payload Examples

The G.718 payload structure enables flexible transport either by

carrying all layers in the same payload or separating the layers into
separate payloads. The following subsections illustrate different
possibilities for transport by simple examples. Note that examples
do not show the full payload structure to keep the illustration
simple.

A.1. Simple Payload Examples

A.1.1. All The Layers in The Same Payload

The illustration below shows layers L1-L3 from two encoded frames
encapsulated into separate payloads using single transport block.

+-------+--------+-----+------+------+------+
| RTP1 | L-ID=3 |NF=0 |F1-L1 |F1-L2 |F1-L3 |
+-------+--------+-----+------+------+------+

+-------+--------+-----+------+------+------+
| RTP2 | L-ID=3 |NF=0 |F2-L1 |F2-L2 |F2-L3 |
+-------+--------+-----+------+------+------+

In the case where the same layers from two input frames are
encapsulated into one payload using single transport block, the
structure is as shown below.

+-------+--------+-----+------+------+------+------+------+------+
| RTP1 | L-ID=3 |NF=1 |F1-L1 |F2-L1 |F1-L2 |F2-L2 |F3-L3 |F2-L3 |
+-------+--------+-----+------+------+------+------+------+------+
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The third example illustrates the case where the layers L1-L3 from
two input frames are encapsulated into one payload using two separate
transport blocks, the first one carrying L1 and the other one
containing L2 and L3.

+-------+--------+-----+------+------+
| RTP1 | L-ID=1 |NF=1 |F1-L1 |F2-L1 |
+-------+--------+-----+------+------+------+------+
| L-ID=7 |NF=1 |F1-L2 |F2-L2 |F2-L2 |F2-L3 |
+--------+-----+------+------+------+------+

A.1.2. Layers in Seperate RTP Streams

In this case the data for each layer is transmitted in its own
payload.

In the first example each transport block including a single EDU is

carried in its own RTP payload.

+-------+--------+-----+-----+ +-------+--------+-----+-----+
| RTP1a | L-ID=1 |NF=0 |F1-L1| | RTP1b | L-ID=6 |NF=0 |F1-L2|
+-------+--------+-----+-----+ +-------+--------+-----+-----+

+-------+--------+-----+-----+ +-------+--------+-----+-----+
| RTP1c |L-ID=10 |NF=0 |F1-L3| | RTP2a | L-ID=1 |NF=0 |F2-L1|
+-------+--------+-----+-----+ +-------+--------+-----+-----+

+-------+--------+-----+-----+ +-------+--------+-----+-----+
| RTP2b | L-ID=6 |NF=0 |F2-L2| | RTP2c |L-ID=10 |NF=0 |F2-L3|
+-------+--------+-----+-----+ +-------+--------+-----+-----+

If the payloads carry data from two consecutive input frames, the
same encoded data as in the previous example is arranged as follows.

+-------+--------+-----+-----+-----+
| RTP1a | L-ID=1 |NF=1 |F1-L1|F2-L1|
+-------+--------+-----+-----+-----+

+-------+--------+-----+-----+-----+
| RTP1b | L-ID=6 |NF=1 |F1-L2|F2-L2|
+-------+--------+-----+-----+-----+

+-------+--------+-----+-----+-----+
| RTP1c |L-ID=10 |NF=1 |F1-L3|F2-L3|
+-------+--------+-----+-----+-----+
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A.2. Advanced Examples

A.2.1. Different Update Rate for Subset of Layers

An example employing different update rates (i.e. different number of

frames per packet) for selected subsets of layers. In these examples
all core codec layers L1-L5 are shown.

+-------+--------+-----+-----+-----+-----+-----+
| RTP1 | L-ID=1 |NF=3 |F1-L1|F2-L1|F3-L1|F4-L1|
+-------+--------+-----+-----+-----+-----+-----+

+-------+--------+-----+-----+-----+-----+-----+
| RTP2a | L-ID=7 |NF=1 |F1-L2|F2-L2|F1-L3|F2-L3|
+-------+--------+-----+-----+-----+-----+-----+

+-------+--------+-----+-----+-----+
| RTP3a |L-ID=14 |NF=0 |F1-L4|F1-L5|
+-------+--------+-----+-----+-----+

+-------+--------+-----+-----+-----+
| RTP3b |L-ID=14 |NF=0 |F2-L4|F2-L5|
+-------+--------+-----+-----+-----+

+-------+--------+-----+-----+-----+-----+-----+
| RTP2b | L-ID=7 |NF=1 |F3-L2|F4-L2|F3-L3|F4-L3|
+-------+--------+-----+-----+-----+-----+-----+

+-------+--------+-----+-----+-----+
| RTP3c |L-ID=14 |NF=0 |F3-L4|F3-L5|
+-------+--------+-----+-----+-----+

+-------+--------+-----+-----+-----+
| RTP3d |L-ID=14 |NF=0 |F4-L4|F4-L5|
+-------+--------+-----+-----+-----+
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A.2.2. Redundant Frames With Limited Set of Layers

An example transmitting layers L1-L3 as primary data and L1 (of the

previous frame) as redundant data is shown below. Each payload
carries one primary (i.e. new) frame in one transport block and one
redundant frame, which in this example is the frame preceding the
primary frame, in another transport block.

+-------+--------+-----+-----+--------+-----+-----+-----+-----+
| RTP1 | L-ID=1 |NF=0 |F0-L1| L-ID=3 |NF=0 |F1-L1|F1-L2|F1-L3|
+-------+--------+-----+-----+--------+-----+-----+-----+-----+

+-------+--------+-----+-----+--------+-----+-----+-----+-----+
| RTP2 | L-ID=1 |NF=0 |F1-L1| L-ID=3 |NF=0 |F2-L1|F2-L2|F2-L3|
+-------+--------+-----+-----+--------+-----+-----+-----+-----+

+-------+--------+-----+-----+--------+-----+-----+-----+-----+
| RTP3 | L-ID=1 |NF=0 |F2-L1| L-ID=3 |NF=0 |F3-L1|F3-L2|F3-L3|
+-------+--------+-----+-----+--------+-----+-----+-----+-----+

Alternatively, the payload carrying also redundant data for a subset

of layers can be arranged differently, as shown in the example below.

+-------+--------+-----+-----+-----+-----+--------+-----+-----+
| RTP1 | L-ID=3 |NF=0 |F0-L1|F0-L2|F0-L3| L-ID=1 |NF=0 |F1-L1|
+-------+--------+-----+-----+-----+-----+--------+-----+-----+

+-------+--------+-----+-----+-----+-----+--------+-----+-----+
| RTP2 | L-ID=3 |NF=0 |F1-L1|F1-L2|F1-L3| L-ID=1 |NF=0 |F2-L1|
+-------+--------+-----+-----+-----+-----+--------+-----+-----+

+-------+--------+-----+-----+-----+-----+--------+-----+-----+
| RTP3 | L-ID=3 |NF=0 |F2-L1|F2-L2|F2-L3| L-ID=1 |NF=0 |F3-L1|
+-------+--------+-----+-----+-----+-----+--------+-----+-----+

Now the first transport block carries the primary data and the second
transport block carries the redundant data, which in this case covers
the frame following the primary frame. The benefit of this approach
is that the redundant data is included in the last (secondary)
transport block of the payload, which might be beneficial for
possible payload scaling operation within the network.
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Authors' Addresses

Glen Zorn (editor)

Network Zen
227/358 Thanon Sanphawut
Bang Na, Bangkok 10260
Thailand

Phone: +66 (0) 87-040-4617

EMail: [email protected]

Ye-Kui Wang
Qualcomm Incorporated
5775 Morehouse Drive
San Diego, CA 92121
USA

Phone: +1-858-651-8345
EMail: [email protected]

Ari Lakaniemi
Independent Contributor

EMail: [email protected]
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