RTP G.711.1: RTP Payload Format for ITU-T Recommendation G.711.1
How to put wideband G.711.1 voice into RTP packets, with a one-byte header that says which quality mode follows.
Sollaud, A., "RTP Payload Format for ITU-T Recommendation G.711.1", RFC 5391, DOI 10.17487/RFC5391, November 2008, <https://www.rfc-editor.org/info/rfc5391>.
@misc{rfc5391,
series = {Request for Comments},
number = 5391,
howpublished = {RFC 5391},
publisher = {RFC Editor},
doi = {10.17487/RFC5391},
url = {https://www.rfc-editor.org/info/rfc5391},
author = {A. Sollaud},
title = {{RTP Payload Format for ITU-T Recommendation G.711.1}},
pagetotal = 14,
year = 2008,
month = nov,
} RFC text
Contents
Network Working Group A. Sollaud
Request for Comments: 5391 France Telecom
Category: Standards Track November 2008
RTP Payload Format for ITU-T Recommendation G.711.1
Status of This Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (c) 2008 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents (http://trustee.ietf.org/
license-info) in effect on the date of publication of this document.
Please review these documents carefully, as they describe your rights
and restrictions with respect to this document.
Abstract
This document specifies a Real-time Transport Protocol (RTP) payload
format to be used for the ITU Telecommunication Standardization
Sector (ITU-T) G.711.1 audio codec. Two media type registrations are
also included.
Table of Contents
1. Introduction ....................................................2
2. Background ......................................................2
3. RTP Header Usage ................................................3
4. Payload Format ..................................................4
4.1. Payload Header .............................................4
4.2. Audio Data .................................................5
5. Payload Format Parameters .......................................6
5.1. PCMA-WB Media Type Registration ............................7
5.2. PCMU-WB Media Type Registration ............................8
5.3. Mapping to SDP Parameters ..................................9
5.3.1. Offer-Answer Model Considerations ...................9
5.3.2. Declarative SDP Considerations .....................11
6. G.711 Interoperability .........................................11
7. Congestion Control .............................................12
8. Security Considerations ........................................12
9. IANA Considerations ............................................12
10. References ....................................................13
10.1. Normative References .....................................13
10.2. Informative References ...................................13
1. Introduction
The ITU Telecommunication Standardization Sector (ITU-T)
Recommendation G.711.1 [ITU-G.711.1] is an embedded wideband
extension of the Recommendation G.711 [ITU-G.711] audio codec. This
document specifies a payload format for packetization of G.711.1
encoded audio signals into the Real-time Transport Protocol (RTP).
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. Background
G.711.1 is a G.711 embedded wideband speech and audio coding
algorithm operating at 64, 80, and 96 kbps. At 64 kbps, G.711.1 is
fully interoperable with G.711. Hence, an efficient deployment in
existing G.711-based Voice over IP (VoIP) infrastructures is
foreseen.
The codec operates on 5-ms frames, and the default sampling rate is
16 kHz. Input and output at 8 kHz are also supported for narrowband
modes.
The encoder produces an embedded bitstream structured in three layers
corresponding to three available bit rates: 64, 80, and 96 kbps. The
bitstream can be truncated at the decoder side or by any component of
the communication system to adjust, "on the fly", the bit rate to the
desired value.
The following table gives more details on these layers.
+-------+------------------------+----------+
| Layer | Description | Bit rate |
+-------+------------------------+----------+
| L0 | G.711 compatible | 64 kbps |
| L1 | narrowband enhancement | 16 kbps |
| L2 | wideband enhancement | 16 kbps |
+-------+------------------------+----------+
Table 1: Layers description
The combinations of these three layers results in the definition of
four modes, as per the following table.
+------+----+----+----+------------+----------+
| Mode | L0 | L1 | L2 | Audio band | Bit rate |
+------+----+----+----+------------+----------+
| R1 | x | | | narrowband | 64 kbps |
| R2a | x | x | | narrowband | 80 kbps |
| R2b | x | | x | wideband | 80 kbps |
| R3 | x | x | x | wideband | 96 kbps |
+------+----+----+----+------------+----------+
Table 2: Modes description
3. RTP Header Usage
The format of the RTP header is specified in [RFC3550]. The payload
format defined in this document uses the fields of the header in a
manner consistent with that specification.
marker (M):
G.711.1 does not define anything specific regarding Discontinuous
Transmission (DTX), a.k.a. silence suppression. Codec-independent
mechanisms may be used, like the generic comfort-noise payload
format defined in [RFC3389].
For applications that send either no packets or occasional
comfort-noise packets during silence, the first packet of a
talkspurt -- that is, the first packet after a silence period
during which packets have not been transmitted contiguously --
SHOULD be distinguished by setting the marker bit in the RTP data
header to one. The marker bit in all other packets is zero. The
beginning of a talkspurt MAY be used to adjust the playout delay
to reflect changing network delays. Applications without silence
suppression MUST set the marker bit to zero.
payload type (PT):
The assignment of an RTP payload type for this 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
codec or specify that the payload type is to be bound dynamically
(see Section 5.3).
timestamp:
The RTP timestamp clock frequency is the same as the default
sampling frequency: 16 kHz.
G.711.1 has also the capability to operate with 8-kHz sampled
input/output signals. It does not affect the bitstream, and the
decoder does not require a priori knowledge about the sampling
rate of the original signal at the input of the encoder.
Therefore, depending on the implementation and the audio acoustic
capabilities of the devices, the input of the encoder and/or the
output of the decoder can be configured at 8 kHz; however, a
16-kHz RTP clock rate MUST always be used.
The duration of one frame is 5 ms, corresponding to 80 samples at
16 kHz. Thus, the timestamp is increased by 80 for each
consecutive frame.
4. Payload Format
The complete payload consists of a payload header of 1 octet,
followed by one or more consecutive G.711.1 audio frames of the same
mode.
The mode may change between packets, but not within a packet.
4.1. Payload Header
The payload header is illustrated below.
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|0 0 0 0 0| MI |
+-+-+-+-+-+-+-+-+
The five most significant bits are reserved for further extension and
MUST be set to zero and MUST be ignored by receivers.
The Mode Index (MI) field (3 bits) gives the mode of the following
frame(s) as per the table:
+------------+--------------+------------+
| Mode Index | G.711.1 mode | Frame size |
+------------+--------------+------------+
| 1 | R1 | 40 octets |
| 2 | R2a | 50 octets |
| 3 | R2b | 50 octets |
| 4 | R3 | 60 octets |
+------------+--------------+------------+
Table 3: Modes in payload header
All other values of MI are reserved for future use and MUST NOT be
used.
Payloads received with an undefined MI value MUST be discarded.
If a restricted mode-set has been set up by the signaling (see
Section 5), payloads received with an MI value not in this set MUST
be discarded.
4.2. Audio Data
After this payload header, the consecutive audio frames are packed in
order of time, that is, oldest first. All frames MUST be of the same
mode, indicated by the MI field of the payload header.
Within a frame, layers are always packed in the same order: L0 then
L1 for mode R2a, L0 then L2 for mode R2b, L0 then L1 then L2 for mode
R3. This is illustrated below.
+-------------------------------+
R1 | L0 |
+-------------------------------+
+-------------------------------+--------+
R2a | L0 | L1 |
+-------------------------------+--------+
+-------------------------------+--------+
R2b | L0 | L2 |
+-------------------------------+--------+
+-------------------------------+--------+--------+
R3 | L0 | L1 | L2 |
+-------------------------------+--------+--------+
The size of one frame is given by the mode, as per Table 3, and the
actual number of frames is easy to infer from the size of the audio
data part:
nb_frames = (size_of_audio_data) / (size_of_one_frame).
Only full frames must be considered. So if there is a remainder to
the division above, the corresponding remaining bytes in the received
payload MUST be ignored.
5. Payload Format Parameters
This section defines the parameters that may be used to configure
optional features in the G.711.1 RTP transmission.
Both A-law and mu-law G.711 are supported in the core layer L0, but
there is no interoperability between A-law and mu-law. So two media
types with the same parameters will be defined: audio/PCMA-WB for
A-law core, and audio/PCMU-WB for mu-law core. This is consistent
with the audio/PCMA and audio/PCMU media types separation for G.711
audio.
The parameters are defined here as part of the media subtype
registrations for the G.711.1 codec. A 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 appropriate format used with that control protocol.
5.1. PCMA-WB Media Type Registration
This registration is done using the template defined in [RFC4288] and
following [RFC4855].
Type name: audio
Subtype name: PCMA-WB
Required parameters: none
Optional parameters:
mode-set: restricts the active codec mode set to a subset of all
modes. Possible values are a comma-separated list of modes
from the set: 1, 2, 3, 4 (see Mode Index in Table 3 of RFC
5391). The modes are listed in order of preference; first is
preferred. If mode-set is specified, frames encoded with modes
outside of the subset MUST NOT be sent in any RTP payload. If
not present, all codec modes are allowed.
ptime: the recommended length of time (in milliseconds)
represented by the media in a packet. It should be an integer
multiple of 5 ms (the frame size). See Section 6 of RFC 4566.
maxptime: the maximum length of time (in milliseconds) that can
be encapsulated in a packet. It should be an integer multiple
of 5 ms (the frame size). See Section 6 of RFC 4566.
Encoding considerations: This media type is framed and contains
binary data. See Section 4.8 of RFC 4288.
Security considerations: See Section 8 of RFC 5391.
Interoperability considerations: none
Published specification: RFC 5391
Applications that use this media type: Audio and video conferencing
tools.
Additional information: none
Person & email address to contact for further information: Aurelien
Sollaud, aurelien.sollaud@orange-ftgroup.com
Intended usage: COMMON
Restrictions on usage: This media type depends on RTP framing, and
hence is only defined for transfer via RTP.
Author: Aurelien Sollaud
Change controller: IETF Audio/Video Transport working group delegated
from the IESG
5.2. PCMU-WB Media Type Registration
This registration is done using the template defined in [RFC4288] and
following [RFC4855].
Type name: audio
Subtype name: PCMU-WB
Required parameters: none
Optional parameters:
mode-set: restricts the active codec mode-set to a subset of all
modes. Possible values are a comma-separated list of modes
from the set: 1, 2, 3, 4 (see Mode Index in Table 3 of RFC
5391). The modes are listed in order of preference; first is
preferred. If mode-set is specified, frames encoded with modes
outside of the subset MUST NOT be sent in any RTP payload. If
not present, all codec modes are allowed.
ptime: the recommended length of time (in milliseconds)
represented by the media in a packet. It should be an integer
multiple of 5 ms (the frame size). See Section 6 of RFC 4566.
maxptime: the maximum length of time (in milliseconds) that can
be encapsulated in a packet. It should be an integer multiple
of 5 ms (the frame size). See Section 6 of RFC 4566.
Encoding considerations: This media type is framed and contains
binary data. See Section 4.8 of RFC 4288.
Security considerations: See Section 8 of RFC 5391.
Interoperability considerations: none
Published specification: RFC 5391
Applications that use this media type: Audio and video conferencing
tools.
Additional information: none
Person & email address to contact for further information: Aurelien
Sollaud, aurelien.sollaud@orange-ftgroup.com
Intended usage: COMMON
Restrictions on usage: This media type depends on RTP framing, and
hence is only defined for transfer via RTP.
Author: Aurelien Sollaud
Change controller: IETF Audio/Video Transport working group delegated
from the IESG
5.3. Mapping to SDP Parameters
The information carried in the media type specification has a
specific mapping to fields in the Session Description Protocol (SDP)
[RFC4566], which is commonly used to describe RTP sessions. When SDP
is used to specify sessions employing the G.711.1 codec, the mapping
is as follows:
o The media type ("audio") goes in SDP "m=" as the media name.
o The media subtype ("PCMA-WB" or "PCMU-WB") goes in SDP "a=rtpmap"
as the encoding name. The RTP clock rate in "a=rtpmap" MUST be
16000 for G.711.1.
o The parameter "mode-set" goes in the SDP "a=fmtp" attribute by
copying it as a "mode-set=<value>" string.
o The parameters "ptime" and "maxptime" go in the SDP "a=ptime" and
"a=maxptime" attributes, respectively.
5.3.1. Offer-Answer Model Considerations
The following considerations apply when using SDP offer-answer
procedures [RFC3264] to negotiate the use of G.711.1 payload in RTP:
o Since G.711.1 is an extension of G.711, the offerer SHOULD
announce G.711 support in its "m=audio" line, with G.711.1
preferred. This will allow interoperability with both G.711.1 and
G.711-only capable parties. This is done by offering the PCMA
media subtype in addition to PCMA-WB, and/or PCMU in addition to
PCMU-WB.
Below is an example part of such an offer, for A-law:
m=audio 54874 RTP/AVP 96 8
a=rtpmap:96 PCMA-WB/16000
a=rtpmap:8 PCMA/8000
As a reminder, the payload format for G.711 is defined in Section
4.5.14 of [RFC3551].
o The "mode-set" parameter is bi-directional; i.e., the restricted
mode-set applies to media both to be received and sent by the
declaring entity. If a mode-set was supplied in the offer, the
answerer MUST return either the same mode-set or a subset of this
mode-set. The answerer MAY change the preference order. If no
mode-set was supplied in the offer, the anwerer MAY return a mode-
set to restrict the possible modes. In any case, the mode-set in
the answer then applies for both offerer and answerer. The
offerer MUST NOT send frames of a mode that has been removed by
the answerer.
For multicast sessions, if "mode-set" is supplied in the offer,
the answerer SHALL only participate in the session if it supports
the offered mode-set.
o The parameters "ptime" and "maxptime" will in most cases not
affect interoperability. The SDP offer-answer handling of the
"ptime" parameter is described in [RFC3264]. The "maxptime"
parameter MUST be handled in the same way.
o Any unknown parameter in an offer MUST be ignored by the receiver
and MUST NOT be included in the answer.
Below are some example parts of SDP offer-answer exchanges.
o Example 1
Offer: G.711.1 all modes, with G.711 fallback, prefers mu-law
m=audio 54874 RTP/AVP 96 97 0 8
a=rtpmap:96 PCMU-WB/16000
a=rtpmap:97 PCMA-WB/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
Answer: all modes accepted, both mu- and A-law.
m=audio 59452 RTP/AVP 96 97
a=rtpmap:96 PCMU-WB/16000
a=rtpmap:97 PCMA-WB/16000
o Example 2
Offer: G.711.1 all modes, with G.711 fallback, prefers A-law
m=audio 54874 RTP/AVP 96 97 8 0
a=rtpmap:96 PCMA-WB/16000
a=rtpmap:97 PCMU-WB/16000
Answer: wants only A-law mode R3
m=audio 59452 RTP/AVP 96
a=rtpmap:96 PCMA-WB/16000
a=fmtp:96 mode-set=4
o Example 3
Offer: G.711.1 A-law with two modes, R2b and R3, with R3 preferred
m=audio 54874 RTP/AVP 96
a=rtpmap:96 PCMA-WB/16000
a=fmtp:96 mode-set=4,3
Answer: accepted
m=audio 59452 RTP/AVP 96
a=rtpmap:96 PCMA-WB/16000
a=fmtp:96 mode-set=4,3
If the answerer had wanted to restrict to one mode, it would have
answered with only one value in the mode-set, for example mode-
set=3 for mode R2b.
5.3.2. Declarative SDP Considerations
For declarative use of SDP, nothing specific is defined for this
payload format. The configuration given by the SDP MUST be used when
sending and/or receiving media in the session.
6. G.711 Interoperability
The L0 layer of G.711.1 is fully interoperable with G.711, and is
embedded in all modes of G.711.1. This provides an easy G.711.1 -
G.711 transcoding process.
A gateway or any other network device receiving a G.711.1 packet can
easily extract a G.711-compatible payload, without the need to decode
and re-encode the audio signal. It simply has to take the audio data
of the payload, and strip the upper layers (L1 and/or L2), if any.
If a G.711.1 packet contains several frames, the concatenation of the
L0 layers of each frame will form a G.711-compatible payload.
7. Congestion Control
Congestion control for RTP SHALL be used in accordance with [RFC3550]
and any appropriate profile (for example, [RFC3551]).
The embedded nature of G.711.1 audio data can be helpful for
congestion control, since a coding mode with a lower bit rate can be
selected when needed. This property is usable only when multiple
modes have been negotiated (either no "mode-set" parameter in the SDP
or a "mode-set" with at least two modes).
The number of frames encapsulated in each RTP payload influences the
overall bandwidth of the RTP stream, due to the header overhead.
Packing more frames in each RTP payload can reduce the number of
packets sent and hence the header overhead, at the expense of
increased delay and reduced error robustness.
8. Security Considerations
RTP packets using the payload format defined in this specification
are subject to the general security considerations discussed in the
RTP specification [RFC3550] and any appropriate profile (for example,
[RFC3551]).
As this format transports encoded speech/audio, the main security
issues include confidentiality, integrity protection, and
authentication of the speech/audio itself. The payload format itself
does not have any built-in security mechanisms. Any suitable
external mechanisms, such as the Secure Real-time Transport Protocol
(SRTP) [RFC3711], MAY be used.
This payload format and the G.711.1 encoding do not exhibit any
significant non-uniformity in the receiver-end computational load,
and thus they are unlikely to pose a denial-of-service threat due to
the receipt of pathological datagrams. In addition, they do not
contain any type of active content such as scripts.
9. IANA Considerations
Two new media subtypes (audio/PCMA-WB and audio/PCMU-WB) have been
registered by IANA. See Sections 5.1 and 5.2.
10. References
10.1. Normative References
[ITU-G.711.1] International Telecommunications Union, "Wideband
embedded extension for G.711 pulse code modulation",
ITU-T Recommendation G.711.1, March 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.
[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.
[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.
[RFC4855] Casner, S., "Media Type Registration of RTP Payload
Formats", RFC 4855, February 2007.
10.2. Informative References
[ITU-G.711] International Telecommunications Union, "Pulse code
modulation (PCM) of voice frequencies", ITU-T
Recommendation G.711, November 1988.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload
for Comfort Noise (CN)", RFC 3389, September 2002.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and
K. Norrman, "The Secure Real-time Transport Protocol
(SRTP)", RFC 3711, March 2004.
Author's Address
Aurelien Sollaud
France Telecom
2 avenue Pierre Marzin
Lannion Cedex 22307
France
Phone: +33 2 96 05 15 06
EMail: aurelien.sollaud@orange-ftgroup.com
AI ELI5
On this page
§1Introduction
ITU-T Recommendation G.711.1 is an embedded wideband extension of the G.711 audio codec. RFC 5391 specifies how G.711.1 encoded audio is packetized into the Real-time Transport Protocol (RTP).
The document also registers two media types. It uses the RFC 2119 key words, so MUST and SHOULD carry their usual force.
A new, better-sounding kind of phone audio needs a standard way to ride inside Internet packets. This RFC is that recipe.
§2The G.711.1 Codec: Layers and Modes
G.711.1 operates at 64, 80 and 96 kbps. At 64 kbps it is fully interoperable with G.711, which is why an efficient deployment in existing G.711-based VoIP infrastructures is foreseen. The codec works on 5-ms frames and its default sampling rate is 16 kHz. Input and output at 8 kHz are also supported for narrowband modes.
The encoder produces an embedded bitstream in three layers: L0 (G.711 compatible, 64 kbps), L1 (narrowband enhancement, 16 kbps) and L2 (wideband enhancement, 16 kbps). The bitstream can be truncated by the decoder or by any component of the communication system to adjust the bit rate on the fly.
Combining layers gives four modes: R1 (L0, narrowband, 64 kbps), R2a (L0 and L1, narrowband, 80 kbps), R2b (L0 and L2, wideband, 80 kbps) and R3 (all three layers, wideband, 96 kbps).
The sound comes in layers, like a sandwich. The bottom layer is the basic old-style voice, and extra layers add clearer sound on top. You can eat the sandwich with fewer layers if you need to.
§3RTP Header Usage
The RTP header follows RFC 3550. G.711.1 defines nothing specific for silence suppression (DTX), so codec-independent mechanisms such as the comfort-noise format of RFC 3389 may be used. For applications that send no packets or occasional comfort-noise packets during silence, the first packet of a talkspurt SHOULD have the marker bit set to one. Applications without silence suppression MUST set it to zero.
The payload type assignment is outside the scope of this document; the RTP profile is expected to assign one or make it dynamic.
The timestamp clock runs at 16 kHz. Even if the encoder input or decoder output is configured at 8 kHz, a 16-kHz RTP clock rate MUST always be used. One frame lasts 5 ms, which is 80 samples at 16 kHz, so the timestamp increases by 80 for each consecutive frame.
The packet's label has a clock that always ticks at the same fast speed, even if the recording was made more slowly. Each 5-ms slice of sound moves the clock forward by 80 ticks.
§4The Payload Format
The payload is a payload header of 1 octet, followed by one or more consecutive G.711.1 frames of the same mode. The mode may change between packets, but not within a packet.
The five most significant bits of the header are reserved: they MUST be set to zero and MUST be ignored by receivers. The remaining 3 bits are the Mode Index (MI). MI 1 means R1 with 40-octet frames, 2 means R2a with 50 octets, 3 means R2b with 50 octets and 4 means R3 with 60 octets. Other MI values are reserved and MUST NOT be used; payloads with an undefined MI value MUST be discarded, as MUST payloads whose MI is outside a mode-set restricted by signaling.
Frames are packed oldest first. Within a frame the layers are always in the same order: L0 then L1 for R2a, L0 then L2 for R2b, and L0, L1, L2 for R3. The receiver finds the frame count by dividing the audio data size by the frame size for the mode. Only full frames count, and any remainder bytes MUST be ignored.
Each package starts with a sticker saying which quality level is inside. After that come the sound slices, oldest first, each the same size. Any crumbs left over at the end get thrown away.
| Field | Name | Size | Value | Description |
|---|---|---|---|---|
RESERVED | Reserved | 5 bits | 00000 | Must be zero and ignored by receivers. |
MI | MI | 3 bits | Mode Index of the frames that follow. |
- 1R1 — Frame size 40 octets.
- 2R2a — Frame size 50 octets.
- 3R2b — Frame size 50 octets.
- 4R3 — Frame size 60 octets.
§5Media Types and SDP Negotiation
Both A-law and mu-law G.711 can form the core layer, and the two are not interoperable. So the RFC registers audio/PCMA-WB (A-law core) and audio/PCMU-WB (mu-law core), with identical optional parameters: mode-set, ptime and maxptime. The mode-set is a comma-separated list drawn from 1, 2, 3, 4, in order of preference, and frames in other modes MUST NOT be sent. If it is absent, all modes are allowed. The ptime and maxptime values should be multiples of 5 ms.
In SDP, the subtype goes in a=rtpmap with a clock rate that MUST be 16000, and mode-set is copied into a=fmtp.
In offer/answer (RFC 3264), the offerer SHOULD also offer plain PCMA or PCMU, with G.711.1 preferred. The mode-set applies in both directions. If the offer has one, the answerer MUST return the same set or a subset of it, and MAY change the order. The offerer MUST NOT send a mode the answerer removed. Unknown parameters in an offer MUST be ignored and left out of the answer. For declarative SDP, nothing specific is defined.
Before a call, both sides agree which quality levels they will use, like two friends agreeing which language to speak. If one friend only knows the old language, the call still works.
- Offer: PCMA-WB and PCMU-WB, G.711 fallbackNo mode-set, so all modes are allowed.
- Answer: PCMA-WB, mode-set=4Wants only A-law mode R3.
§6Working with Plain G.711
The L0 layer of G.711.1 is fully interoperable with G.711 and is embedded in every G.711.1 mode. A gateway can therefore take the audio data of a packet and strip the upper layers (L1 and/or L2), with no need to decode and re-encode the audio. If a packet holds several frames, joining the L0 layers of each frame forms a G.711-compatible payload.
Because the basic layer is the same as the old format, a helper in the middle can just peel off the extra layers and pass the plain voice along.
§7Congestion Control and Packing Frames
Congestion control for RTP SHALL be used in accordance with RFC 3550 and any appropriate profile such as RFC 3551. The embedded design helps: a lower bit rate mode can be selected when needed, but only if several modes were negotiated (no mode-set, or a mode-set with at least two modes).
The number of frames per packet also affects bandwidth because of header overhead. Packing more frames reduces the packet count and overhead, at the cost of more delay and less robustness to errors.
If the road is jammed, the sender can switch to a smaller layer stack. Putting many slices in one box saves boxes, but you wait longer to fill it and a lost box loses more.
§8Security Considerations
Packets are subject to the general security considerations of RFC 3550 and the profile in use. The main issues are confidentiality, integrity protection and authentication of the audio itself. The payload format has no built-in security mechanisms, but suitable external ones, such as SRTP (RFC 3711), MAY be used.
The format and the encoding do not show significant non-uniformity in receiver computational load, so they are unlikely to pose a denial-of-service threat from pathological datagrams, and they contain no active content such as scripts.
The package has no lock of its own, so if the call is private you need to add a lock from outside. Opening the package is not risky, though, because it holds only sound.