WDIFF

draft-ietf-avt-rfc2793bis-07.txt --> draft-ietf-avt-rfc2793bis-08.txt

Network Working Group G. Hellstrom
Internet Draft Omnitor AB
<draft-ietf-avt-rfc2793bis-07.txt>
<draft-ietf-avt-rfc2793bis-08.txt> P. Jones
Expires: December 2004 January 2005 Cisco Systems, Inc.
June
July 2004

RTP Payload for Text Conversation

Status of this Memo

This document is an Internet-Draft and is subject to all provisions
of Section 10 of RFC2026.

By submitting this Internet-Draft, we certify that any applicable
patent or other IPR claims of which we are aware have been
disclosed, and any of which we become aware will be disclosed, in
accordance with RFC 3668 (BCP 79).

By submitting this Internet-Draft, we accept the provisions of
Section 3 of RFC 3667 (BCP 78).

Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet-
Drafts.

Internet-Drafts are draft documents valid for a maximum of six
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This document is a submission of the IETF AVT WG. Comments should
be directed to the AVT WG mailing list, avt@ietf.org.

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.

Two payload formats are described. One for transmitting text on a
separate RTP session dedicated for the transmission of text, and

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one for transmitting audio and text data within one 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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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 text/t140 Data..........5
3.3 Payload Format for Transmission of audio/t140c Data........5
3.4 The "T140block"............................................5
3.5 Synchronization of Text with Other Media...................6 Media...................5
3.6 Synchronization considerations for the audio/t140c format..6
3.7 RTP packet header..........................................7
4. Protection against loss of data................................8 data................................7
4.1 Payload Format when using Redundancy.......................8
4.2 Using redundancy with the text/t140 format.................8
4.3 Using redundancy with the audio/t140c format...............9
5. Recommended Procedure.........................................10 Procedure..........................................9
5.1 Recommended Basic Procedure...............................10
5.2 Transmission before and after "Silent Periods"............10
5.3 Detection of Lost Text Packets............................11
5.4 Compensation for Packets Out of Order.....................12
6. Parameter for Character Transmission Rate.....................12
7. Examples......................................................13
7.1 RTP Packetization Examples for the text/t140 format.......13
7.2 RTP Packetization Examples for the audio/t140c format.....15
7.3 SDP Examples..............................................17 Examples..............................................16
8. Security Considerations.......................................18 Considerations.......................................17
8.1 Confidentiality...........................................18 Confidentiality...........................................17
8.2 Integrity.................................................18 Integrity.................................................17
8.3 Source authentication.....................................18 authentication.....................................17
9. Congestion Considerations.....................................19 Considerations.....................................18
10. IANA considerations..........................................20 considerations..........................................19
10.1 Registration of MIME Media Type text/t140................20 text/t140................19
10.2 Registration of MIME Media Type audio/t140c..............21 audio/t140c..............20
10.3 SDP mapping of MIME parameters...........................22 parameters...........................21
10.4 Offer/Answer Consideration...............................23 Consideration...............................21
11. Authors' Addresses...........................................23 Addresses...........................................22
12. Acknowledgements.............................................23 Acknowledgements.............................................22
13. Normative References.........................................22
14. Informative References.......................................23
15. Intellectual Property Statement..............................23
16. Copyright Statement..........................................24

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[Notes to RFC Editor:
1. All references to RFC XXXX are to be replaced by references to
the RFC number of this memo, when published.
2. All references to RFC YYYY [9] are to be replaced by references
to the document that registers the text/red MIME type.]

1. Introduction

This document defines two payload types 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

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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

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network translation devices. This, in turn, eases the design and
increases the possibility for prompt and proper media delivery.

This document obsoletes RFC 2793 [16]. The text clarifies
ambiguities in RFC 2793, improves on the specific implementation
requirements learned through development experience, gives explicit
usage examples, and introduces a method of transporting text
interleaved with voice within the same RTP session.

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

Two payload formats for real-time text transmission with RTP [2]
are described in this memo, one for general text conversation use,
called text/t140 after its MIME registration, and another for use
between PSTN gateways called audio/t140c.

3.1 Motivations and rationale

The text/t140 format is intended to be used for text transmitted on
a separate RTP session dedicated for the transmission of text and
not shared with other media.

Devices implementing payloads according to this memo MUST support
text/t140. The text/t140 format SHALL be used for any non-gateway
application. It MAY also be used between transit gateways. It MAY
be used simultaneously with other media streams, transmitted as a
separate RTP session, as required in real time multimedia
applications.

The text/t140 format specified in this memo is compatible with its
earlier definition in RFC2793. It is just refined, with the main
intention to minimize interoperability problems and encourage good
reliability and functionality.

By specifying text transmission as a text medium, many good effects
are gained. Routing, device selection, invocation of transcoding,
selection of quality of service parameters and other high and low
level functions are depending on each medium being explicitly
specified.

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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 DTMF is extracted
and transmitted within an RTP session [15].

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The audio/t140c format is OPTIONAL and SHALL NOT be used for other
applications 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 is a limited and scarce
resource.

3.2 Payload Format for Transmission of text/t140 Data

A text/t140 conversation RTP payload format consists of one and
only one block of T.140 data, referred to as a "T140block" (see
section 3.4). There are no additional headers specific to this
payload format. The fields in the RTP header are set as defined in
section 3.7.

3.3 Payload Format for Transmission of audio/t140c Data

An audio/t140c conversation RTP payload format consists of a 16-bit
"T140block counter" with bit and carried in network byte order according to normal
habits in IP communication, as specified in rfc 791[12] (see RFC 791 [12]
Annex B, B), followed by one and only one "T140block" (see section
3.4). The fields in the RTP header are set as defined in section
3.7.

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
only refers 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.

3.4 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

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[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.

3.5 Synchronization of Text with Other Media

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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 RTCP SDES function.
It is dependent on the particular application and is outside the
scope of this document.

3.6 Synchronization considerations for the audio/t140c format.

When audio/t140c is used, it is 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. Note, Note also, 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.

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.

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The audio/t140c format allows simultaneous audio and text
transmission only if each stream is given its own SSRC. When using
the same SSRC, the expectation is that at each moment, only one
payload type in

Note that stream is selected for play-out. It is
RECOMMENDED to use the same SSRC for all cases when audio and text
are not intended 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 SSRC and sending its own RTP and RTCP (such an
end point will appear to others in the session as two participants
with different SSRC, but the same RTCP SDES CNAME). Further details

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around using multiple payloads in an RTP session can be used simultaneously. found in
RFC 3550 [2].

3.7 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
types "text/t140" and "audio/t140c" (see section 10). If
redundancy is used per RFC 2198, another payload type number
needs to be provided for the redundancy format. MIME types for
identifying RFC 2198 are available in RFC 3555 and RFC YYYY [9].

Sequence number: The definition of sequence numbers is available in
RFC 3550 [2]. When transmitting text using the payload format for
text/t140, it is used for detection of packet loss and packets
out of order, and can be used in the process of retrieval of
redundant text, reordering of text and marking missing text.
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. A clock frequency of
1000 Hz MUST be used for text/t140. 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 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.

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4. Protection against loss of data

A method MUST

Consideration must be used for ensuring that devoted to keeping loss of text caused by
packet loss is kept within acceptable limits. ( See (See ITU-T F.700 Annex 3.
[17])

The default method that MUST be used when no other method is
explicitly selected is redundancy in accordance with RFC 2198 [3].

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When this method is used, the original text and two redundant
generations SHOULD be transmittted, transmitted if the application or end-to-
end 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. If end-to-end
network conditions allow, text data MAY be sent without additional
protection.

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 text/t140 or 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. The exact payload format
is slightly different for the text/t140 format and for the
audio/t140c format.

Redundant data that would need a timestamp offset higher than 16383
due to its age at transmission MUST NOT be included in transmitted
packets.

4.2 Using redundancy with the text/t140 format.

Since text is transmitted only when there is text to transmit, the
timestamp is not used to identify a lost packet. Rather, missing
sequence numbers are used to detect lost text packets at reception.
Also, since sequence numbers are not provided in the redundant
header, some additional rules must be followed to allow the

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redundant data corresponding to missing primary data to be merged
properly into the stream of primary data T140blocks when using the
text/t140 payload format. They are:

- Each redundant data block MUST contain the same data as a
T140block previously transmitted as primary data.
- The redundant data MUST be placed in age order with most
recent redundant T140block last in the redundancy area.

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- All T140blocks from the oldest desired generation up through
the generation immediately preceding the new (primary)
T140block MUST be included.

These rules allow the sequence numbers for the redundant T140blocks
to be inferred by counting backwards from the sequence number in
the RTP header. The result will be that all the text in the
payload will be contiguous and in order.

If there is a gap in the received RTP sequence numbers for
text/t140, and redundant T140blocks are available in a subsequent
packet, the sequence numbers for the redundant T140blocks should be
inferred by counting backwards from the sequence number in the RTP
header for that packet. If there are redundant T140blocks with
sequence numbers matching those that are missing, the redundant
T140blocks may be substituted for the missing T140blocks.

4.3 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 merged retrieved and used properly into the stream of
received primary T140block data
T140blocks when using the audio/t140c payload format.

Each

All non-empty redundant data block 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.

For the audio/t140c payload format, this rule allows the T140block
counters for the redundant T140blocks to be retrieved.

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.

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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.

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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.

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 "Silent 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.

In the text/t140 format, an empty T140block contains no data.
Likewise with audio/t140c, an

An empty T140block contains no text and
the T140block counter MUST NOT be present.

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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, while
the remainder of the redundancy packet will contain empty
T140blocks.

When using the text/t140 payload format, any empty T140block that
is sent as primary data MUST be included as redundant T140blocks in
subsequent packets just as normal text T140blocks would be, unless
the empty T140block is too old to be transmitted. This is done so
that sequence number inference for the redundant T140blocks will be
correct, as explained in section 4.2.

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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

Packet loss for text/t140 packets MAY be detected by observing gaps
in the sequence numbers of RTP packets received by the receiver.

With audio/t140c, however, packets following a text packet might be
audio packets of a format other than audio/t140c, so from observing
sequence number gaps it is not possible to tell what medium was
lost. Rather, receivers detect the loss of an audio/t140c packet by
observing the value of the T140block counter in a subsequent
audio/t140c packet.

With text/t140 the loss of packets can also be is usually detected by
comparison of the sequence numbers in the RTCP reports with of RTP packets
sent and received. as they arrive. Any
discrepancy MAY be used to detect indicate loss. The highest RTP sequence
number received may also be compared with that in RTCP reports, as
an additional check for loss of the last packet before an idle
period.

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].

Since empty T140blocks are transmitted in the beginning of an idle
period, there is a slight risk of falsely marking loss of text,
when only an empty T140block was lost and when using text/t140.
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

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loss. False detection will be avoided when using audio/t140c by
observing the value of the T140block counter value.

If redundancy is used with the text/t140 format, and a packet is
received with fewer redundancy levels than normally in the session,
it SHOULD be treated as if one empty T140block has been received
for each excluded level in the received packet. This is because the
only occasion when a T140block is excluded from transmission is
when it is an empty T140block that has become too old to be
transmitted.

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.

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The text/t140 format relies on use of the sequence number in the
RTP packet header for detection of loss and is therefore not
suitable for an application where it needs to be alternating with
other payloads in the same RTP stream. It would be complicated and
unreliable to try to detect loss of data at the edges of the shifts
between t140 text and other stream contents. It is therefore
RECOMMENDED to be the only payload type in the RTP stream.

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 which 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

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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 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.

Examples of use in SDP are found in section 7.3.

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In receipt of this parameter, devices MUST adhere to the request by
transmitting characters at a rate at or below the specified
<integer> value. Note that this parameter was not defined in RFC
2793 [16]. Therefore implementations of the text/t140 format may be
in use that do not recognize and act according to this parameter.
Receivers of text/t140 SHALL therefore be designed so that they can
handle temporary reception of characters at a higher rate than this
parameter specifies, so that malfunction because of buffer overflow
is avoided for text conversation with human input.

7. Examples

7.1 RTP Packetization Examples for the text/t140 format.

Below is an example of a text/t140 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| T140 PT | sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp (1000Hz) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
| T.140 encoded data +
| |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

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Below is an example of a text/t140 RTP packet with one redundant
T140block.

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| T140 PT | timestamp offset of "R" | "R" block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| T140 PT | "R" T.140 encoded redundant data |
+-+-+-+-+-+-+-+-+ +
| |
+ +---------------+
|
+ | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +-+-+-+-+-+
| "P" T.140 encoded primary data |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Below is an example of an RTP packet with one redundant T140block

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using text/t140 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.

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| |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

As a follow-on to the previous example, the example below shows
the next RTP packet in the sequence which does contain a real
T140block when using the text/t140 payload format. Note that the
empty block is present in the redundant transmissions of the
text/t140 payload format. This example shows 2 levels of
redundancy and one primary data block. The value of the "R2
block length" would be set to zero in order to
represent the empty T140block.
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| T140 PT | timestamp offset of "R2" | "R2" block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| T140 PT | timestamp offset of "R1" | "R1" block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| T140 PT | "R1" T.140 encoded redundant data |
+-+-+-+-+-+-+-+-+ +---------------+
| | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +-+-+-+
| "P" T.140 encoded primary data |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

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7.2 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 |

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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 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 |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

As a follow-on to the previous example, the example below shows
the next RTP packet in the sequence which 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

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audio/t140c payload format, any previously transmitted "empty"
T140blocks are NOT included as redundant data in subsequent
packets.

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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 |
+-+-+-+-+-+-+-+-+ +
| |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

7.3 SDP Examples

Below is an example of SDP describing RTP text transport on port
11000:

m=text 11000 RTP/AVP 98
a=rtpmap:98 t140/1000

Below is an example of SDP similar to the above example, but also
utilizing RFC 2198 to provide the recommended two levels of
redundancy for the text packets:

m=text 11000 RTP/AVP 98 100
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98

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

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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:

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. SRTP [14] 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 [14] 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

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established, SRTP [14] 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 the section about congestion in
chapter 2 of RFC 3551 [11] 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 [13], 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 are not sufficient, the only remaining action is to
terminate the session.

As guidance, some load figures are provided here.

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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 octets UTF-8 characters, no header
compression, two redundant levels and 300 ms between transmissions,
the maximum load of this application is 3300 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 two RTP payloads named "t140" and "t140c" and
two associated MIME types, "text/t140" and "audio/t140c", to be
registered by IANA.

10.1 Registration of MIME Media Type text/t140

MIME media type name: text

MIME subtype name: t140

Required parameters:
rate: The RTP timestamp clock rate, which is equal to the
sampling rate. The only valid value is 1000.

Optional parameters:
cps: The maximum number of characters that may be received
per second. The deafult value is 30.

Encoding considerations: T.140 text can be transmitted with RTP
as specified in RFC XXXX.

Security considerations: See section 8 of RFC XXXX.

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Interoperability considerations: This format is the same as
specified in RFC2793. For RFC2793 the "cps=" parameter was not
defined. Therefore there may be implementations that do not

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consider this parameter. Receivers need to take that into
account.

Published specification: ITU-T T.140 Recommendation.
RFC XXXX.

Applications which use this media type:
Text communication terminals and text conferencing tools.

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:
Gunnar Hellstrom
E-mail: gunnar.hellstrom@omnitor.se

Intended usage: COMMON

Author / Change controller:
Gunnar Hellstrom | IETF avt WG
gunnar.hellstrom@omnitor.se |

10.2 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 deafult value is 30.

Encoding considerations: T.140 text can be transmitted with RTP
as specified in RFC XXXX.

Security considerations: See section 8 of RFC XXXX.

Interoperability considerations: None

Published specification: ITU-T T.140 Recommendation.
RFC XXXX.

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Interoperability considerations: None

Published specification: ITU-T T.140 Recommendation.
RFC XXXX.

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
paulej@packetizer.com |

10.3 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 text/t140 or
audio/t140c format, the mapping is as follows:

- The MIME type ("text") goes in SDP "m=" as the media name.

- The MIME subtype (payload format name) goes in SDP "a=rtpmap"
as the encoding name. The RTP clock rate in "a=rtpmap" MUST be
1000 for text/t140. 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 YYYY [9] and RFC 2198 [3].

10.4 Offer/Answer Consideration

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In order to achieve interoperability within the framework of the
offer/answer model [10], 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. 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
E-mail: 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
E-mail: paulej@packetizer.com

12. 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, to 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.

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13. 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", RFC
3550, July 2003.

[3] Perkins, C., Kouvelas, I., Hardman, V., Handley, M. and J.
Bolot, "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.

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[5] ISO/IEC 10646-1: (1993), Universal Multiple Octet Coded
Character Set.

[6] Yergeau, F., "UTF-8, a transformation format of ISO 10646",
RFC 3629, December 2003.

[7] Handley, M., Jacobson, V., "SDP: Session Description
Protocol", RFC 2327, April 1998.

[8] Rosenberg, J., Schulzrinne, H., "An RTP Payload Format for
Generic Forward Error Correction", RFC 2733, December 1999.

[9] Jones, P. , "Registration of the text/red MIME Sub-Type",
draft-ietf-avt-text-red, RFC YYYY, 2004.

[10] Rosenberg, J., Schulzrinne, H., "An Offer/Answer Model with
the Session Description Protocol (SDP)", RFC 3264, June 2002.

[11] Schultzrinne, J., Perkins, C., "RTP Profile for Audio and
Video Conference with Minimal Control", RFC 3551, July 2003.

[12] Postel, J.,"Internet Protocol", RFC 791, 1981.

14. Informative References

[13] Floyd, S., Kempf, J., IAB Concerns Regarding Congestion
Control for Voice Traffic in the Internet, RFC 3714,March 2004

[14] Baugher, McGrew, Carrara, Naslund, Norrman, The Secure Real-
Time Transport Protocol (SRTP), RFC 3711, March 2004.

[15] Schulzrinne, H., Petrack, S., "RTP Payload for DTMF Digits,
Telephony Tones and Telephony Signals", RFC 2833, May 2000.

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[16] Hellstrom, G., "RTP Payload for text conversation.", RFC2793,
2000

[17] ITU-T Recommendation F.700, Annex 3. Framework Recommendation
for multimedia services, Nov 2000.

15. Intellectual Property Statement

The IETF takes no position regarding the validity or scope of any
Intellectual Property Rights or other rights that might be claimed
to pertain to the implementation or use of the technology described
in this document or the extent to which any license under such
rights might or might not be available; nor does it represent that
it has made any independent effort to identify any such rights.
Information on the IETF's procedures with respect to rights in IETF
Documents can be found in RFC 3667 (BCP 78) and RFC 3668 (BCP 79).

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Copies of IPR disclosures made to the IETF Secretariat and any
assurances of licenses to be made available, or the result of an
attempt made to obtain a general license or permission for the use
of such proprietary rights by implementers or users of this
specification can be obtained from the IETF on-line IPR repository
at http://www.ietf.org/ipr.

The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights that may cover technology that may be required to implement
this standard. Please address the information to the IETF at
ietf-ipr@ietf.org.

16. Copyright Statement

This document is subject to the rights, licenses and restrictions
contained in BCP 78, and except as set forth therein, the authors
retain all their rights.

Disclaimer of Validity

This document and the information contained herein are provided on
an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE
REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND
THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES,
EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT
THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR
ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A
PARTICULAR PURPOSE.

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----