Real-Time Transport Protocol (RTP) Payload for Text Conversation Interleaved in an Audio Stream
RFC 4351
| Document | Type | RFC - Historic (January 2006; No errata) | |
|---|---|---|---|
| Authors | Paul Jones , Gunnar Hellstrom | ||
| Last updated | 2018-12-20 | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | plain text html pdf htmlized bibtex | ||
| Stream | WG state | (None) | |
| Document shepherd | No shepherd assigned | ||
| IESG | IESG state | RFC 4351 (Historic) | |
| Action Holders |
(None)
|
||
| Consensus Boilerplate | Unknown | ||
| Telechat date | |||
| Responsible AD | Allison Mankin | ||
| Send notices to | csp@csperkins.org, magnus.westerlund@ericsson.com | ||
Network Working Group G. Hellstrom
Request for Comments: 4351 Omnitor AB
Category: Historic P. Jones
Cisco Systems, Inc.
January 2006
Real-Time Transport Protocol (RTP) Payload for
Text Conversation Interleaved in an Audio Stream
Status of This Memo
This memo defines a Historic Document for the Internet community. It
does not specify an Internet standard of any kind. Distribution of
this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2006).
Abstract
This memo describes how to carry real-time text conversation session
contents in RTP packets. Text conversation session contents are
specified in ITU-T Recommendation T.140.
One payload format is described for transmitting audio and text data
within a single RTP session.
This RTP payload description recommends a method to include redundant
text from already transmitted packets in order to reduce the risk of
text loss caused by packet loss.
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RFC 4351 RTP Payload for Text in an Audio Stream January 2006
Table of Contents
1. Introduction ....................................................3
2. Conventions Used in This Document ...............................4
3. Usage of RTP ....................................................4
3.1. Motivations and Rationale ..................................4
3.2. Payload Format for Transmission of audio/t140c Data ........4
3.3. The "T140block" ............................................5
3.4. Synchronization of Text with Other Media ...................5
3.5. Synchronization Considerations for the audio/t140c Format ..5
3.6. RTP Packet Header ..........................................6
4. Protection against Loss of Data .................................7
4.1. Payload Format When Using Redundancy .......................7
4.2. Using Redundancy with the audio/t140c Format ...............8
5. Recommended Procedure ...........................................8
5.1. Recommended Basic Procedure ................................8
5.2. Transmission before and after "Idle Periods" ...............9
5.3. Detection of Lost Text Packets .............................9
5.4. Compensation for Packets Out of Order .....................10
6. Parameter for Character Transmission Rate ......................10
7. Examples .......................................................11
7.1. RTP Packetization Examples for the audio/t140c Format .....11
7.2. SDP Examples ..............................................12
8. Security Considerations ........................................13
8.1. Confidentiality ...........................................13
8.2. Integrity .................................................13
8.3. Source Authentication .....................................13
9. Congestion Considerations ......................................14
10. IANA Considerations ...........................................15
10.1. Registration of MIME Media Type audio/t140c ..............15
10.2. SDP Mapping of MIME Parameters ...........................16
10.3. Offer/Answer Consideration ...............................17
11. Acknowledgements ..............................................17
12. Normative References ..........................................17
13. Informative References ........................................18
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1. Introduction
This document defines a payload type for carrying text conversation
session contents in RTP [2] packets. Text conversation session
contents are specified in ITU-T Recommendation T.140 [1]. Text
conversation is used alone or in connection to other conversational
facilities, such as video and voice, to form multimedia conversation
services. Text in multimedia conversation sessions is sent
character-by-character as soon as it is available, or with a small
delay for buffering.
The text is intended to be entered by human users from a keyboard,
handwriting recognition, voice recognition, or any other input
method. The rate of character entry is usually at a level of a few
characters per second or less. In general, only one or a few new
characters are expected to be transmitted with each packet. Small
blocks of text may be prepared by the user and pasted into the user
interface for transmission during the conversation, occasionally
causing packets to carry more payload.
T.140 specifies that text and other T.140 elements must be
transmitted in ISO 10646-1[5] code with UTF-8 [6] transformation.
That makes it easy to implement internationally useful applications
and to handle the text in modern information technology environments.
The payload of an RTP packet following this specification consists of
text encoded according to T.140 without any additional framing. A
common case will be a single ISO 10646 character, UTF-8 encoded.
T.140 requires the transport channel to provide characters without
duplication and in original order. Text conversation users expect
that text will be delivered with no or a low level of lost
information.
Therefore a mechanism based on RTP is specified here. It gives text
arrival in correct order, without duplication, and with detection and
indication of loss. It also includes an optional possibility to
repeat data for redundancy to lower the risk of loss. Since packet
overhead is usually much larger than the T.140 contents, the increase
in bandwidth with the use of redundancy is minimal.
By using RTP for text transmission in a multimedia conversation
application, uniform handling of text and other media can be achieved
in, as examples, conferencing systems, firewalls, and network
translation devices. This, in turn, eases the design and increases
the possibility for prompt and proper media delivery.
This document introduces a method of transporting text interleaved
with voice within the same RTP session.
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2. Conventions Used in This Document
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 RFC 2119 [4].
3. Usage of RTP
The payload format for real-time text transmission with RTP [2]
described in this memo is intended for use between Public Switched
Telephone Network (PSTN) gateways and is called audio/t140c.
3.1. Motivations and Rationale
The audio/t140c payload specification is intended to allow gateways
that are interconnecting two PSTN networks to interleave, through a
single RTP session, audio and text data received on the PSTN circuit.
This is comparable to the way in which dual-tone multifrequency
(DTMF) is extracted and transmitted within an RTP session [14].
The audio/t140c format SHALL NOT be used for applications other than
PSTN gateway applications. In such applications, a specific
profiling document MAY make it REQUIRED for a specific application.
The reason to prefer to use audio/t140c could be for gateway
application where the ports are a limited and scarce resource.
Applications SHOULD use RFC 4103 [15] for real-time text
communication that falls outside the limited scope of this
specification.
3.2. Payload Format for Transmission of audio/t140c Data
An audio/t140c conversation RTP payload format consists of a 16-bit
"T140block counter" carried in network byte order (see RFC 791 [11]
Annex B), followed by one and only one "T140block" (see section 3.3).
The fields in the RTP header are set as defined in section 3.6.
The T140block counter MUST be initialized to zero the first time that
a packet containing a T140block is transmitted and MUST be
incremented by 1 each time that a new block is transmitted. Once the
counter reaches the value 0xFFFF, the counter is reset to 0 the next
time the counter is incremented. This T140block counter is used to
detect lost blocks and to avoid duplication of blocks.
For the purposes of readability, the remainder of this document
refers only to the T140block without making explicit reference to the
T140block counter. Readers should understand that when using the
audio/t140c format, the T140block counter MUST always precede the
actual T140block, including redundant data transmissions.
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3.3. The "T140block"
T.140 text is UTF-8 coded as specified in T.140 with no extra
framing. The T140block contains one or more T.140 code elements as
specified in [1]. Most T.140 code elements are single ISO 10646 [5]
characters, but some are multiple-character sequences. Each
character is UTF-8 encoded [6] into one or more octets. Each block
MUST contain an integral number of UTF-8-encoded characters
regardless of the number of octets per character. Any composite
character sequence (CCS) SHOULD be placed within one block.
3.4. Synchronization of Text with Other Media
Usually, each medium in a session utilizes a separate RTP stream. As
such, if synchronization of the text and other media packets is
important, the streams MUST be associated when the sessions are
established and the streams MUST share the same reference clock
(refer to the description of the timestamp field as it relates to
synchronization in section 5.1 of RFC 3550). Association of RTP
streams can be done through the CNAME field of RTP Control Protocol
(RTCP) SDES function. It is dependent on the particular application
and is outside the scope of this document.
3.5. Synchronization Considerations for the audio/t140c Format
The audio/t140c packets are generally transmitted as interleaved
packets between voice packets or other kinds of audio packets with
the intention to create one common audio signal in the receiving
equipment to be used for alternating between text and voice. The
audio/t140c payload is then used to play out audio signals according
to a PSTN textphone coding method (usually a modem).
One should observe the RTP timestamps of the voice, text, or other
audio packets in order to reproduce the stream correctly when playing
out the audio. Also, note that incoming text from a PSTN circuit
might be at a higher bit-rate than can be played out on an egress
PSTN circuit. As such, it is possible that, on the egress side, a
gateway may not complete the play out of the text packets before it
is time to play the next voice packet. Given that this application
is primarily for the benefit of users of PSTN textphone devices, it
is strongly RECOMMENDED that all received text packets be properly
reproduced on the egress gateway before considering any other
subsequent audio packets.
If necessary, voice and other audio packets should be discarded in
order to properly reproduce the text signals on the PSTN circuit,
even if the text packets arrive late.
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The PSTN textphone users commonly use turn-taking indicators in the
text stream, so it can be expected that as long as text is
transmitted, it is valid text and should be given priority over
voice.
Note that the usual RTP semantics apply with regards to switching
payload formats within an RTP session. A sender MAY switch between
"audio/t140c" and some other format within an RTP session, but MUST
NOT send overlapping data using two different audio formats within an
RTP session. This does not prohibit an implementation from being
split into two logical parts to send overlapping data, each part
using a different synchronization source (SSRC) and sending its own
RTP and RTCP (such an endpoint will appear to others in the session
as two participants with different SSRCs, but the same RTCP SDES
CNAME). Further details around using multiple payloads in an RTP
session can be found in RFC 3550 [2].
3.6. RTP Packet Header
Each RTP packet starts with a fixed RTP header. The following fields
of the RTP fixed header are specified for T.140 text streams:
Payload Type (PT): The assignment of an RTP payload type is specific
to the RTP profile under which this payload format is used. For
profiles that use dynamic payload type number assignment, this
payload format can be identified by the MIME type "audio/t140c"
(see section 10). If redundancy is used per RFC 2198, another
payload type number needs to be provided for the redundancy
format. The MIME type for identifying RFC 2198 is available in
RFC 3555 [17].
Sequence number: The definition of sequence numbers is available in
RFC 3550 [2]. Character loss is detected through the T140block
counter when using the audio/t140c payload format.
Timestamp: The RTP Timestamp encodes the approximate instance of
entry of the primary text in the packet. For audio/t140c, the
clock frequency MAY be set to any value, and SHOULD be set to the
same value as for any audio packets in the same RTP stream in
order to avoid RTP timestamp rate switching. The value SHOULD be
set by out of band mechanisms. Sequential packets MUST NOT use
the same timestamp. Since packets do not represent any constant
duration, the timestamp cannot be used to directly infer packet
loss.
M-bit: The M-bit MUST be included. The first packet in a session,
and the first packet after an idle period, SHOULD be distinguished
by setting the marker bit in the RTP data header to one. The
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marker bit in all other packets MUST be set to zero. The
reception of the marker bit MAY be used for refined methods for
detection of loss.
4. Protection against Loss of Data
Consideration must be devoted to keeping loss of text caused by
packet loss within acceptable limits. (See ITU-T F.703 [16].)
The default method that MUST be used when no other method is
explicitly selected is redundancy in accordance with RFC 2198 [3].
When this method is used, the original text and two redundant
generations SHOULD be transmitted if the application or end-to-end
conditions do not call for other levels of redundancy to be used.
Other protection methods MAY be used. Forward Error Correction
mechanisms as per RFC 2733 [8] or any other mechanism with the
purpose of increasing the reliability of text transmission MAY be
used as an alternative or complement to redundancy. Text data MAY be
sent without additional protection if end-to-end network conditions
allow the text quality requirements specified in ITU-T F.703 [16] to
be met in all anticipated load conditions.
4.1. Payload Format When Using Redundancy
When using the format with redundant data, the transmitter may select
a number of T140block generations to retransmit in each packet. A
higher number introduces better protection against loss of text but
marginally increases the data rate.
The RTP header is followed by one or more redundant data block
headers, one for each redundant data block to be included. Each of
these headers provides the timestamp offset and length of the
corresponding data block plus a payload type number indicating the
payload format audio/t140c.
After the redundant data block headers follows the redundant data
fields carrying T140blocks from previous packets, and finally the new
(primary) T140block for this packet.
Redundant data that would need a timestamp offset higher than 16383
due to its age at transmission MUST NOT be included in transmitted
packets.
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4.2. Using Redundancy with the audio/t140c Format
Since sequence numbers are not provided in the redundant header and
since the sequence number space is shared by all audio payload types
within an RTP session, a sequence number in the form of a T140block
counter is added to the T140block for transmission. This allows the
redundant T140block data corresponding to missing primary data to be
retrieved and used properly into the stream of received T140block
data when using the audio/t140c payload format.
All non-empty redundant data blocks MUST contain the same data as a
T140block previously transmitted as primary data, and be identified
with a T140block counter equating to the original T140block counter
for that T140block.
The T140block counters preceding the text in the T140block enables
the ordering by the receiver. If there is a gap in the T140block
counter value of received audio/t140c packets, and if there are
redundant T140blocks with T140block counters matching those that are
missing, the redundant T140blocks may be substituted for the missing
T140blocks.
The value of the length field in the redundant header indicates the
length of the concatenated T140block counter and the T140block.
5. Recommended Procedure
This section contains RECOMMENDED procedures for usage of the payload
format. Based on the information in the received packets, the
receiver can:
- reorder text received out of order.
- mark where text is missing because of packet loss.
- compensate for lost packets by using redundant data.
5.1. Recommended Basic Procedure
Packets are transmitted when there is valid T.140 data to transmit.
T.140 specifies that T.140 data MAY be buffered for transmission with
a maximum buffering time of 500 ms. A buffering time of 300 ms is
RECOMMENDED when the application or end-to-end network conditions are
not known to require another value.
If no new data is available for a longer period than the buffering
time, the transmission process is in an idle period.
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When new text is available for transmission after an idle period, it
is RECOMMENDED to send it as soon as possible. After this
transmission, it is RECOMMENDED to buffer T.140 data in buffering
time intervals until next idle period. This is done in order to keep
the maximum bit-rate usage for text at a reasonable level. The
buffering time MUST be selected so that text users will perceive a
real-time text flow.
5.2. Transmission before and after "Idle Periods"
When valid T.140 data has been sent and no new T.140 data is
available for transmission after the selected buffering time, an
empty T140block SHOULD be transmitted. This situation is regarded to
be the beginning of an idle period. The procedure is recommended in
order to more rapidly detect potentially missing text before an idle
period or when the audio stream switches from the transmission of
audio/t140c to some other form of audio.
An empty T140block contains no data, neither T.140 data nor a
T140block counter.
When redundancy is used, transmission continues with a packet at
every transmission timer expiration and insertion of an empty
T.140block as primary, until the last non-empty T140block has been
transmitted as primary and as redundant data with all intended
generations of redundancy. The last packet before an idle period
will contain only one non-empty T140block as redundant data, and the
empty primary T140block.
When using the audio/t140c payload format, empty T140blocks sent as
primary data SHOULD NOT be included as redundant T140blocks, as it
would simply be a waste of bandwidth to send them and it would
introduce a risk of false detection of loss.
After an idle period, the transmitter SHOULD set the M-bit to one in
the first packet with new text.
5.3. Detection of Lost Text Packets
Receivers detect the loss of an audio/t140c packet by observing the
value of the T140block counter in a subsequent audio/t140c packet.
Missing data SHOULD be marked by insertion of a missing text marker
in the received stream for each missing T140block, as specified in
ITU-T T.140 Addendum 1 [1].
Procedures based on detection of the packet with the M-bit set to one
MAY be used to reduce the risk for introducing false markers of loss.
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False detection will also be avoided when using audio/t140c by
observing the value of the T140block counter value.
If two successive packets have the same number of redundant
generations, it SHOULD be treated as the general redundancy level for
the session. Change of the general redundancy level SHOULD only be
done after an idle period.
5.4. Compensation for Packets Out of Order
For protection against packets arriving out of order, the following
procedure MAY be implemented in the receiver. If analysis of a
received packet reveals a gap in the sequence and no redundant data
is available to fill that gap, the received packet SHOULD be kept in
a buffer to allow time for the missing packet(s) to arrive. It is
RECOMMENDED that the waiting time be limited to 1 second.
If a packet with a T140block belonging to the gap arrives before the
waiting time expires, this T140block is inserted into the gap and
then consecutive T140blocks from the leading edge of the gap may be
consumed. Any T140block that does not arrive before the time limit
expires should be treated as lost and a missing text marker inserted
(see section 5.3).
6. Parameter for Character Transmission Rate
In some cases, it is necessary to limit the rate at which characters
are transmitted. For example, when a PSTN gateway is interworking
between an IP device and a PSTN textphone, it may be necessary to
limit the character rate from the IP device in order to avoid
throwing away characters in case of buffer overflow at the PSTN
gateway.
To control the character transmission rate, the MIME parameter "cps"
in the "fmtp" attribute [7] is defined (see section 10). It is used
in Session Description Protocol (SDP) with the following syntax:
a=fmtp:<format> cps=<integer>
The <format> field is populated with the payload type that is used
for text. The <integer> field contains an integer representing the
maximum number of characters that may be received per second. The
value shall be used as a mean value over any 10-second interval. The
default value is 30.
In receipt of this parameter, devices MUST adhere to the request by
transmitting characters at a rate at or below the specified <integer>
value. Examples of use in SDP are found in section 7.2.
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7. Examples
7.1. RTP Packetization Examples for the audio/t140c Format
Below is an example of an audio/t140c RTP packet without redundancy.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC=0 |M| T140c PT | sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp (8000Hz) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| T140block counter | T.140 encoded data |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Below is an example of an RTP packet with one redundant T140block
using audio/t140c payload format. The primary data block is empty,
which is the case when transmitting a packet for the sole purpose of
forcing the redundant data to be transmitted in the absence of any
new data. Note that since this is the audio/t140c payload format,
the redundant block of T.140 data is immediately preceded with a
T140block counter.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC=0 |M| "RED" PT | sequence number of primary |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp of primary encoding "P" |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| T140c PT | timestamp offset of "R" | "R" block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| T140c PT | "R" T140block counter | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
| "R" T.140 encoded redundant data |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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As a follow-on to the previous example, the example below shows the
next RTP packet in the sequence that does contain a new real
T140block when using the audio/t140c payload format. This example
has 2 levels of redundancy and one primary data block. Since the
previous primary block was empty, no redundant data is included for
that block. This is because when using the audio/t140c payload
format, any previously transmitted "empty" T140blocks are NOT
included as redundant data in subsequent packets.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC=0 |M| "RED" PT | sequence number of primary |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp of primary encoding "P" |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| T140c PT | timestamp offset of "R1" | "R1" block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| T140c PT | "R1" T140block counter | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
| "R1" T.140 encoded redundant data |
+ +---------------+
| | "P" T140block |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| counter | "P" T.140 encoded primary data |
+-+-+-+-+-+-+-+-+ +
| |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
7.2. SDP Examples
Below is an example of SDP describing RTP text interleaved with G.711
audio packets within the same RTP session from port 7200 and at a
maximum text rate of 6 characters per second:
m=audio 7200 RTP/AVP 0 98
a=rtpmap:98 t140c/8000
a=fmtp:98 cps=6
Below is an example using RFC 2198 to provide the recommended two
levels of redundancy to the text packets in an RTP session with
interleaving text and G.711 at a text rate no faster than 20
characters per second:
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m=audio 7200 RTP/AVP 0 98 100
a=rtpmap:98 t140c/8000
a=fmtp:98 cps=20
a=rtpmap:100 red/8000
a=fmtp:100 98/98/98
Note: While these examples utilize the RTP/AVP profile, it is not
intended to limit the scope of this memo to use with only that
profile. Rather, any appropriate profile may be used in conjunction
with this memo.
8. Security Considerations
All of the security considerations from section 14 of RFC 3550 [2]
apply.
8.1. Confidentiality
Since the intention of the described payload format is to carry text
in a text conversation, security measures in the form of encryption
are of importance. The amount of data in a text conversation session
is low, and therefore any encryption method MAY be selected and
applied to T.140 session contents or to the whole RTP packets.
Secure Realtime Transport Protocol (SRTP) [13] provides a suitable
method for ensuring confidentiality.
8.2. Integrity
It may be desirable to protect the text contents of an RTP stream
against manipulation. SRTP [13] provides methods for providing
integrity that MAY be applied.
8.3. Source Authentication
Measures to make sure that the source of text is the intended one can
be accomplished by a combination of methods.
Text streams are usually used in a multimedia control environment.
Security measures for authentication are available and SHOULD be
applied in the registration and session establishment procedures, so
that the identity of the sender of the text stream is reliably
associated with the person or device setting up the session. Once
established, SRTP [13] mechanisms MAY be applied to ascertain that
the source is maintained the same during the session.
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9. Congestion Considerations
The congestion considerations from section 10 of RFC 3550 [2],
section 6 of RFC 2198 [3], and any used profile (e.g., the part about
congestion in section 2 of RFC 3551 [10]) apply with the following
application-specific considerations.
Automated systems MUST NOT use this format to send large amounts of
text at a rate significantly above that which a human user could
enter.
Even if the network load from users of text conversation is usually
very low, for best-effort networks an application MUST monitor the
packet loss rate and take appropriate actions to reduce its sending
rate if this application sends at higher rate than what TCP would
achieve over the same path. The reason is that this application, due
to its recommended usage of two or more redundancy levels, is very
robust against packet loss. At the same time, due to the low bit-
rate of text conversations, if one considers the discussion in RFC
3714 [12], this application will experience very high packet loss
rates before it needs to perform any reduction in the sending rate.
If the application needs to reduce its sending rate, it SHOULD NOT
reduce the number of redundancy levels below the default amount
specified in section 4. Instead, the following actions are
RECOMMENDED in order of priority:
- Increase the shortest time between transmissions described in
section 5.1 from the recommended 300 ms to 500 ms that is the
highest value allowable according to T.140.
- Limit the maximum rate of characters transmitted.
- Increase the shortest time between transmissions to a higher value,
not higher than 5 seconds. This will cause unpleasant delays in
transmission, beyond what is allowed according to T.140, but text
will still be conveyed in the session with some usability.
- Exclude participants from the session.
Please note that if the reduction in bit-rate achieved through the
above measures is not sufficient, the only remaining action is to
terminate the session.
As guidance, some load figures are provided here as examples based on
use of IPv4, including the load from IP, UDP, and RTP headers without
compression.
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- Experience tells that a common mean character transmission rate
during a complete PSTN text telephony session in reality is around
2 characters per second.
- A maximum performance of 20 characters per second is enough even
for voice-to-text applications.
- With the (unusually high) load of 20 characters per second, in a
language that make use of three-octet UTF-8 characters, two
redundant levels, and 300 ms between transmissions, the maximum
load of this application is 3500 bits/s.
- When the restrictions mentioned above are applied, limiting
transmission to 10 characters per second, using 5 s between
transmissions, the maximum load of this application in a language
that uses one octet per UTF-8 character is 300 bits/s.
Note also, that this payload can be used in a congested situation as
a last resort to maintain some contact when audio and video media
need to be stopped. The availability of one low bit-rate stream for
text in such adverse situations may be crucial for maintaining some
communication in a critical situation.
10. IANA Considerations
This document defines one RTP payload format named "t140" and an
associated MIME type "audio/t140c". They have been registered by the
IANA.
10.1. Registration of MIME Media Type audio/t140c
MIME media type name: audio
MIME subtype name: t140c
Required parameters:
rate: The RTP timestamp clock rate, which is equal to the
sampling rate. This parameter SHOULD have the same value as
for any audio codec packets interleaved in the same RTP
stream.
Optional parameters:
cps: The maximum number of characters that may be received
per second. The default value is 30.
Encoding considerations: T.140 text can be transmitted with RTP
as specified in RFC 4351.
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Security considerations: See section 8 of RFC 4351.
Interoperability considerations: None
Published specification: ITU-T T.140 Recommendation.
RFC 4351.
Applications which use this media type:
Text communication systems and text conferencing tools that
transmit text associated with audio and within the same RTP
session as the audio, such as PSTN gateways that transmit
audio and text signals between two PSTN textphone users
over an IP network.
Additional information: This type is only defined for transfer
via RTP.
Magic number(s): None
File extension(s): None
Macintosh File Type Code(s): None
Person & email address to contact for further information:
Paul E. Jones
E-mail: paulej@packetizer.com
Intended usage: COMMON
Author / Change controller:
Paul E. Jones | IETF avt WG delegated from the IESG
paulej@packetizer.com |
10.2. SDP Mapping of MIME Parameters
The information carried in the MIME media type specification has a
specific mapping to fields in the Session Description Protocol (SDP)
[7], which is commonly used to describe RTP sessions. When SDP is
used to specify sessions employing the audio/t140c format, 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. For audio/t140c, the clock rate MAY be set
to any value, and SHOULD be set to the same value as for any
audio packets in the same RTP stream.
- The parameter "cps" goes in SDP "a=fmtp" attribute.
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- When the payload type is used with redundancy according to RFC
2198, the level of redundancy is shown by the number of elements
in the slash-separated payload type list in the "fmtp" parameter
of the redundancy declaration as defined in RFC 2198 [3].
10.3. Offer/Answer Consideration
In order to achieve interoperability within the framework of the
offer/answer model [9], the following consideration should be made:
- The "cps" parameter is declarative. Both sides may provide a
value, which is independent of the other side.
11. Acknowledgements
The authors want to thank Stephen Casner, Magnus Westerlund, and
Colin Perkins for valuable support with reviews and advice on
creation of this document; Mickey Nasiri at Ericsson Mobile
Communication for providing the development environment; Michele
Mizarro for verification of the usability of the payload format for
its intended purpose; and Andreas Piirimets for editing support.
12. Normative References
[1] ITU-T Recommendation T.140 (1998) - Text conversation protocol
for multimedia application, with amendment 1, (2000).
[2] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", STD 64,
RFC 3550, July 2003.
[3] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M.,
Bolot, J., Vega-Garcia, A., and S. Fosse-Parisis, "RTP Payload
for Redundant Audio Data", RFC 2198, September 1997.
[4] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[5] ISO/IEC 10646-1: (1993), Universal Multiple Octet Coded
Character Set.
[6] Yergeau, F., "UTF-8, a transformation format of ISO 10646", STD
63, RFC 3629, November 2003.
[7] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
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[8] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for
Generic Forward Error Correction", RFC 2733, December 1999.
[9] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[10] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
[11] Postel, J., "Internet Protocol", STD 5, RFC 791, September 1981.
13. Informative References
[12] Floyd, S. and J. Kempf, "IAB Concerns Regarding Congestion
Control for Voice Traffic in the Internet", RFC 3714, March
2004.
[13] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
3711, March 2004.
[14] Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF Digits,
Telephony Tones and Telephony Signals", RFC 2833, May 2000.
[15] Hellstrom, G. and P. Jones, "RTP Payload for Text Conversation",
RFC 4103, June 2005.
[16] ITU-T Recommendation F.703, Multimedia Conversational Services,
Nov 2000.
[17] Casner, S. and P. Hoschka, "MIME Type Registration of RTP
Payload Formats", RFC 3555, July 2003.
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RFC 4351 RTP Payload for Text in an Audio Stream January 2006
Authors' Addresses
Gunnar Hellstrom
Omnitor AB
Renathvagen 2
SE-121 37 Johanneshov
Sweden
Phone: +46 708 204 288 / +46 8 556 002 03
Fax: +46 8 556 002 06
EMail: gunnar.hellstrom@omnitor.se
Paul E. Jones
Cisco Systems, Inc.
7025 Kit Creek Rd.
Research Triangle Park, NC 27709
USA
Phone: +1 919 392 6948
EMail: paulej@packetizer.com
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RFC 4351 RTP Payload for Text in an Audio Stream January 2006
Full Copyright Statement
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