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Versions: 00 rfc4351                                            Historic
                                                        IPR declarations
Network Working Group                                      G. Hellstrom
Internet Draft                                               Omnitor AB
<draft-ietf-avt-audio-t140c-00.txt>                            P. Jones
Expires: February 2005                              Cisco Systems, Inc.
                                                            August 2004

    RTP Payload for Text Conversation interleaved in an audio stream

Status of this Memo

   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
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   Internet-Drafts are draft documents valid for a maximum of six
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   The list of current Internet-Drafts can be accessed at

   The list of Internet-Draft Shadow Directories can be accessed at

   This document is a submission of the IETF AVT WG. Comments should
   be directed to the AVT WG mailing list, avt@ietf.org.


   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 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 audio/t140c Data........4
      3.3 The "T140block"............................................4
      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................................6
      4.1 Payload Format when using Redundancy.......................7
      4.2 Using redundancy with the audio/t140c format...............7
   5. Recommended Procedure..........................................8
      5.1 Recommended Basic Procedure................................8
      5.2 Transmission before and after "Idle Periods"...............8
      5.3 Detection of Lost Text Packets.............................9
      5.4 Compensation for Packets Out of Order......................9
   6. Parameter for Character Transmission Rate.....................10
   7. Examples......................................................10
      7.1 RTP Packetization Examples for the audio/t140c format.....10
      7.2 SDP Examples..............................................12
   8. Security Considerations.......................................12
      8.1 Confidentiality...........................................12
      8.2 Integrity.................................................13
      8.3 Source authentication.....................................13
   9. Congestion Considerations.....................................13
   10. IANA considerations..........................................14
      10.1 Registration of MIME Media Type audio/t140c..............15
      10.2 SDP mapping of MIME parameters...........................16
      10.3 Offer/Answer Consideration...............................16
   11. Authors' Addresses...........................................16
   12. Acknowledgements.............................................17
   13. Normative References.........................................17
   14. Informative References.......................................17
   15. Intellectual Property Statement..............................18
   16. Copyright Statement..........................................18

   [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. ]

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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

   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",
   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 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 DTMF is extracted
   and transmitted within an RTP session [14].

   The audio/t140c format 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 are a limited and scarce resource.

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

   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.

3.3 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

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   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 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. 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

   Note that the usual RTP semantics apply with regards to switching
   payload formats within an RTP session. A sender MAY switch between

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   "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
   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

   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.

   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

   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.

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])

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   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

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 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.

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   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.

   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.

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   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. 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 which does not arrive before the

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   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 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 7.2.

   In receipt of this parameter, devices MUST adhere to the request by
   transmitting characters at a rate at or below the specified
   <integer> value.

7. Examples

7.1 RTP Packetization Examples for the audio/t140c format

   Below is an example of an audio/t140c RTP packet without

    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            |

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

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

    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 |

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   |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:

       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]

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

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   be selected and applied to T.140 session contents or to the whole
   RTP packets. SRTP [13] provides a suitable method for ensuring

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.

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 section
   about congestion in chapter 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

   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

   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:

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   - 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

   - 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 as examples based
   on use of IPv4, including the load from IP, UDP and RTP headers
   without compression.

   -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, 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", to be registered by IANA.

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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

      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 XXXX.

      Security considerations: See section 8 of RFC XXXX.

      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       |

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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.

    -  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. Authors' Addresses

   Gunnar Hellstrom
   Omnitor AB
   Renathvagen 2
   SE-121 37 Johanneshov
   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
   Phone: +1 919 392 6948
   E-mail: paulej@packetizer.com

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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.

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.

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

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

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

14. Informative References

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

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

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

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

   [16] ITU-T Recommendation F.703, Multimedia Conversational
       Services, Nov 2000.

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16. Copyright Statement

   Copyright (C) The Internet Society (2004).

   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

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