GENWiki

Premier IT Outsourcing and Support Services within the UK

User Tools

Site Tools


rfc:rfc4351

Network Working Group G. Hellstrom Request for Comments: 4351 Omnitor AB Category: Historic P. Jones

                                                   Cisco Systems, Inc.
                                                          January 2006
           Real-Time Transport Protocol (RTP) Payload for
         Text Conversation Interleaved in an Audio Stream

Status of This Memo

 This memo defines a Historic Document for the Internet community.  It
 does not specify an Internet standard of any kind.  Distribution of
 this memo is unlimited.

Copyright Notice

 Copyright (C) The Internet Society (2006).

Abstract

 This memo describes how to carry real-time text conversation session
 contents in RTP packets.  Text conversation session contents are
 specified in ITU-T Recommendation T.140.
 One payload format is described for transmitting audio and text data
 within a single RTP session.
 This RTP payload description recommends a method to include redundant
 text from already transmitted packets in order to reduce the risk of
 text loss caused by packet loss.

Hellstrom & Jones Historic [Page 1] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

Table of Contents

 1. Introduction ....................................................3
 2. Conventions Used in This Document ...............................4
 3. Usage of RTP ....................................................4
    3.1. Motivations and Rationale ..................................4
    3.2. Payload Format for Transmission of audio/t140c Data ........4
    3.3. The "T140block" ............................................5
    3.4. Synchronization of Text with Other Media ...................5
    3.5. Synchronization Considerations for the audio/t140c Format ..5
    3.6. RTP Packet Header ..........................................6
 4. Protection against Loss of Data .................................7
    4.1. Payload Format When Using Redundancy .......................7
    4.2. Using Redundancy with the audio/t140c Format ...............8
 5. Recommended Procedure ...........................................8
    5.1. Recommended Basic Procedure ................................8
    5.2. Transmission before and after "Idle Periods" ...............9
    5.3. Detection of Lost Text Packets .............................9
    5.4. Compensation for Packets Out of Order .....................10
 6. Parameter for Character Transmission Rate ......................10
 7. Examples .......................................................11
    7.1. RTP Packetization Examples for the audio/t140c Format .....11
    7.2. SDP Examples ..............................................12
 8. Security Considerations ........................................13
    8.1. Confidentiality ...........................................13
    8.2. Integrity .................................................13
    8.3. Source Authentication .....................................13
 9. Congestion Considerations ......................................14
 10. IANA Considerations ...........................................15
    10.1. Registration of MIME Media Type audio/t140c ..............15
    10.2. SDP Mapping of MIME Parameters ...........................16
    10.3. Offer/Answer Consideration ...............................17
 11. Acknowledgements ..............................................17
 12. Normative References ..........................................17
 13. Informative References ........................................18

Hellstrom & Jones Historic [Page 2] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

1. Introduction

 This document defines a payload type for carrying text conversation
 session contents in RTP [2] packets.  Text conversation session
 contents are specified in ITU-T Recommendation T.140 [1].  Text
 conversation is used alone or in connection to other conversational
 facilities, such as video and voice, to form multimedia conversation
 services.  Text in multimedia conversation sessions is sent
 character-by-character as soon as it is available, or with a small
 delay for buffering.
 The text is intended to be entered by human users from a keyboard,
 handwriting recognition, voice recognition, or any other input
 method.  The rate of character entry is usually at a level of a few
 characters per second or less.  In general, only one or a few new
 characters are expected to be transmitted with each packet.  Small
 blocks of text may be prepared by the user and pasted into the user
 interface for transmission during the conversation, occasionally
 causing packets to carry more payload.
 T.140 specifies that text and other T.140 elements must be
 transmitted in ISO 10646-1[5] code with UTF-8 [6] transformation.
 That makes it easy to implement internationally useful applications
 and to handle the text in modern information technology environments.
 The payload of an RTP packet following this specification consists of
 text encoded according to T.140 without any additional framing.  A
 common case will be a single ISO 10646 character, UTF-8 encoded.
 T.140 requires the transport channel to provide characters without
 duplication and in original order.  Text conversation users expect
 that text will be delivered with no or a low level of lost
 information.
 Therefore a mechanism based on RTP is specified here.  It gives text
 arrival in correct order, without duplication, and with detection and
 indication of loss.  It also includes an optional possibility to
 repeat data for redundancy to lower the risk of loss.  Since packet
 overhead is usually much larger than the T.140 contents, the increase
 in bandwidth with the use of redundancy is minimal.
 By using RTP for text transmission in a multimedia conversation
 application, uniform handling of text and other media can be achieved
 in, as examples, conferencing systems, firewalls, and network
 translation devices.  This, in turn, eases the design and increases
 the possibility for prompt and proper media delivery.
 This document introduces a method of transporting text interleaved
 with voice within the same RTP session.

Hellstrom & Jones Historic [Page 3] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

2. Conventions Used in This Document

 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
 document are to be interpreted as described in RFC 2119 [4].

3. Usage of RTP

 The payload format for real-time text transmission with RTP [2]
 described in this memo is intended for use between Public Switched
 Telephone Network (PSTN) gateways and is called audio/t140c.

3.1. Motivations and Rationale

 The audio/t140c payload specification is intended to allow gateways
 that are interconnecting two PSTN networks to interleave, through a
 single RTP session, audio and text data received on the PSTN circuit.
 This is comparable to the way in which dual-tone multifrequency
 (DTMF) is extracted and transmitted within an RTP session [14].
 The audio/t140c format SHALL NOT be used for applications other than
 PSTN gateway applications.  In such applications, a specific
 profiling document MAY make it REQUIRED for a specific application.
 The reason to prefer to use audio/t140c could be for gateway
 application where the ports are a limited and scarce resource.
 Applications SHOULD use RFC 4103 [15] for real-time text
 communication that falls outside the limited scope of this
 specification.

3.2. Payload Format for Transmission of audio/t140c Data

 An audio/t140c conversation RTP payload format consists of a 16-bit
 "T140block counter" carried in network byte order (see RFC 791 [11]
 Annex B), followed by one and only one "T140block" (see section 3.3).
 The fields in the RTP header are set as defined in section 3.6.
 The T140block counter MUST be initialized to zero the first time that
 a packet containing a T140block is transmitted and MUST be
 incremented by 1 each time that a new block is transmitted.  Once the
 counter reaches the value 0xFFFF, the counter is reset to 0 the next
 time the counter is incremented.  This T140block counter is used to
 detect lost blocks and to avoid duplication of blocks.
 For the purposes of readability, the remainder of this document
 refers only to the T140block without making explicit reference to the
 T140block counter.  Readers should understand that when using the
 audio/t140c format, the T140block counter MUST always precede the
 actual T140block, including redundant data transmissions.

Hellstrom & Jones Historic [Page 4] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

3.3. The "T140block"

 T.140 text is UTF-8 coded as specified in T.140 with no extra
 framing.  The T140block contains one or more T.140 code elements as
 specified in [1].  Most T.140 code elements are single ISO 10646 [5]
 characters, but some are multiple-character sequences.  Each
 character is UTF-8 encoded [6] into one or more octets.  Each block
 MUST contain an integral number of UTF-8-encoded characters
 regardless of the number of octets per character.  Any composite
 character sequence (CCS) SHOULD be placed within one block.

3.4. Synchronization of Text with Other Media

 Usually, each medium in a session utilizes a separate RTP stream.  As
 such, if synchronization of the text and other media packets is
 important, the streams MUST be associated when the sessions are
 established and the streams MUST share the same reference clock
 (refer to the description of the timestamp field as it relates to
 synchronization in section 5.1 of RFC 3550).  Association of RTP
 streams can be done through the CNAME field of RTP Control Protocol
 (RTCP) SDES function.  It is dependent on the particular application
 and is outside the scope of this document.

3.5. Synchronization Considerations for the audio/t140c Format

 The audio/t140c packets are generally transmitted as interleaved
 packets between voice packets or other kinds of audio packets with
 the intention to create one common audio signal in the receiving
 equipment to be used for alternating between text and voice.  The
 audio/t140c payload is then used to play out audio signals according
 to a PSTN textphone coding method (usually a modem).
 One should observe the RTP timestamps of the voice, text, or other
 audio packets in order to reproduce the stream correctly when playing
 out the audio.  Also, note that incoming text from a PSTN circuit
 might be at a higher bit-rate than can be played out on an egress
 PSTN circuit.  As such, it is possible that, on the egress side, a
 gateway may not complete the play out of the text packets before it
 is time to play the next voice packet.  Given that this application
 is primarily for the benefit of users of PSTN textphone devices, it
 is strongly RECOMMENDED that all received text packets be properly
 reproduced on the egress gateway before considering any other
 subsequent audio packets.
 If necessary, voice and other audio packets should be discarded in
 order to properly reproduce the text signals on the PSTN circuit,
 even if the text packets arrive late.

Hellstrom & Jones Historic [Page 5] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

 The PSTN textphone users commonly use turn-taking indicators in the
 text stream, so it can be expected that as long as text is
 transmitted, it is valid text and should be given priority over
 voice.
 Note that the usual RTP semantics apply with regards to switching
 payload formats within an RTP session.  A sender MAY switch between
 "audio/t140c" and some other format within an RTP session, but MUST
 NOT send overlapping data using two different audio formats within an
 RTP session.  This does not prohibit an implementation from being
 split into two logical parts to send overlapping data, each part
 using a different synchronization source (SSRC) and sending its own
 RTP and RTCP (such an endpoint will appear to others in the session
 as two participants with different SSRCs, but the same RTCP SDES
 CNAME).  Further details around using multiple payloads in an RTP
 session can be found in RFC 3550 [2].

3.6. RTP Packet Header

 Each RTP packet starts with a fixed RTP header.  The following fields
 of the RTP fixed header are specified for T.140 text streams:
 Payload Type (PT): The assignment of an RTP payload type is specific
    to the RTP profile under which this payload format is used.  For
    profiles that use dynamic payload type number assignment, this
    payload format can be identified by the MIME type "audio/t140c"
    (see section 10).  If redundancy is used per RFC 2198, another
    payload type number needs to be provided for the redundancy
    format.  The MIME type for identifying RFC 2198 is available in
    RFC 3555 [17].
 Sequence number: The definition of sequence numbers is available in
    RFC 3550 [2].  Character loss is detected through the T140block
    counter when using the audio/t140c payload format.
 Timestamp: The RTP Timestamp encodes the approximate instance of
    entry of the primary text in the packet.  For audio/t140c, the
    clock frequency MAY be set to any value, and SHOULD be set to the
    same value as for any audio packets in the same RTP stream in
    order to avoid RTP timestamp rate switching.  The value SHOULD be
    set by out of band mechanisms.  Sequential packets MUST NOT use
    the same timestamp.  Since packets do not represent any constant
    duration, the timestamp cannot be used to directly infer packet
    loss.
 M-bit: The M-bit MUST be included.  The first packet in a session,
    and the first packet after an idle period, SHOULD be distinguished
    by setting the marker bit in the RTP data header to one.  The

Hellstrom & Jones Historic [Page 6] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

    marker bit in all other packets MUST be set to zero.  The
    reception of the marker bit MAY be used for refined methods for
    detection of loss.

4. Protection against Loss of Data

 Consideration must be devoted to keeping loss of text caused by
 packet loss within acceptable limits. (See ITU-T F.703 [16].)
 The default method that MUST be used when no other method is
 explicitly selected is redundancy in accordance with RFC 2198 [3].
 When this method is used, the original text and two redundant
 generations SHOULD be transmitted if the application or end-to-end
 conditions do not call for other levels of redundancy to be used.
 Other protection methods MAY be used.  Forward Error Correction
 mechanisms as per RFC 2733 [8] or any other mechanism with the
 purpose of increasing the reliability of text transmission MAY be
 used as an alternative or complement to redundancy.  Text data MAY be
 sent without additional protection if end-to-end network conditions
 allow the text quality requirements specified in ITU-T F.703 [16] to
 be met in all anticipated load conditions.

4.1. Payload Format When Using Redundancy

 When using the format with redundant data, the transmitter may select
 a number of T140block generations to retransmit in each packet.  A
 higher number introduces better protection against loss of text but
 marginally increases the data rate.
 The RTP header is followed by one or more redundant data block
 headers, one for each redundant data block to be included.  Each of
 these headers provides the timestamp offset and length of the
 corresponding data block plus a payload type number indicating the
 payload format audio/t140c.
 After the redundant data block headers follows the redundant data
 fields carrying T140blocks from previous packets, and finally the new
 (primary) T140block for this packet.
 Redundant data that would need a timestamp offset higher than 16383
 due to its age at transmission MUST NOT be included in transmitted
 packets.

Hellstrom & Jones Historic [Page 7] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

4.2. Using Redundancy with the audio/t140c Format

 Since sequence numbers are not provided in the redundant header and
 since the sequence number space is shared by all audio payload types
 within an RTP session, a sequence number in the form of a T140block
 counter is added to the T140block for transmission.  This allows the
 redundant T140block data corresponding to missing primary data to be
 retrieved and used properly into the stream of received T140block
 data when using the audio/t140c payload format.
 All non-empty redundant data blocks MUST contain the same data as a
 T140block previously transmitted as primary data, and be identified
 with a T140block counter equating to the original T140block counter
 for that T140block.
 The T140block counters preceding the text in the T140block enables
 the ordering by the receiver.  If there is a gap in the T140block
 counter value of received audio/t140c packets, and if there are
 redundant T140blocks with T140block counters matching those that are
 missing, the redundant T140blocks may be substituted for the missing
 T140blocks.
 The value of the length field in the redundant header indicates the
 length of the concatenated T140block counter and the T140block.

5. Recommended Procedure

 This section contains RECOMMENDED procedures for usage of the payload
 format.  Based on the information in the received packets, the
 receiver can:
  1. reorder text received out of order.
  2. mark where text is missing because of packet loss.
  3. 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.

Hellstrom & Jones Historic [Page 8] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

 When new text is available for transmission after an idle period, it
 is RECOMMENDED to send it as soon as possible.  After this
 transmission, it is RECOMMENDED to buffer T.140 data in buffering
 time intervals until next idle period.  This is done in order to keep
 the maximum bit-rate usage for text at a reasonable level.  The
 buffering time MUST be selected so that text users will perceive a
 real-time text flow.

5.2. Transmission before and after "Idle Periods"

 When valid T.140 data has been sent and no new T.140 data is
 available for transmission after the selected buffering time, an
 empty T140block SHOULD be transmitted.  This situation is regarded to
 be the beginning of an idle period.  The procedure is recommended in
 order to more rapidly detect potentially missing text before an idle
 period or when the audio stream switches from the transmission of
 audio/t140c to some other form of audio.
 An empty T140block contains no data, neither T.140 data nor a
 T140block counter.
 When redundancy is used, transmission continues with a packet at
 every transmission timer expiration and insertion of an empty
 T.140block as primary, until the last non-empty T140block has been
 transmitted as primary and as redundant data with all intended
 generations of redundancy.  The last packet before an idle period
 will contain only one non-empty T140block as redundant data, and the
 empty primary T140block.
 When using the audio/t140c payload format, empty T140blocks sent as
 primary data SHOULD NOT be included as redundant T140blocks, as it
 would simply be a waste of bandwidth to send them and it would
 introduce a risk of false detection of loss.
 After an idle period, the transmitter SHOULD set the M-bit to one in
 the first packet with new text.

5.3. Detection of Lost Text Packets

 Receivers detect the loss of an audio/t140c packet by observing the
 value of the T140block counter in a subsequent audio/t140c packet.
 Missing data SHOULD be marked by insertion of a missing text marker
 in the received stream for each missing T140block, as specified in
 ITU-T T.140 Addendum 1 [1].
 Procedures based on detection of the packet with the M-bit set to one
 MAY be used to reduce the risk for introducing false markers of loss.

Hellstrom & Jones Historic [Page 9] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

 False detection will also be avoided when using audio/t140c by
 observing the value of the T140block counter value.
 If two successive packets have the same number of redundant
 generations, it SHOULD be treated as the general redundancy level for
 the session.  Change of the general redundancy level SHOULD only be
 done after an idle period.

5.4. Compensation for Packets Out of Order

 For protection against packets arriving out of order, the following
 procedure MAY be implemented in the receiver.  If analysis of a
 received packet reveals a gap in the sequence and no redundant data
 is available to fill that gap, the received packet SHOULD be kept in
 a buffer to allow time for the missing packet(s) to arrive.  It is
 RECOMMENDED that the waiting time be limited to 1 second.
 If a packet with a T140block belonging to the gap arrives before the
 waiting time expires, this T140block is inserted into the gap and
 then consecutive T140blocks from the leading edge of the gap may be
 consumed.  Any T140block that does not arrive before the time limit
 expires should be treated as lost and a missing text marker inserted
 (see section 5.3).

6. Parameter for Character Transmission Rate

 In some cases, it is necessary to limit the rate at which characters
 are transmitted.  For example, when a PSTN gateway is interworking
 between an IP device and a PSTN textphone, it may be necessary to
 limit the character rate from the IP device in order to avoid
 throwing away characters in case of buffer overflow at the PSTN
 gateway.
 To control the character transmission rate, the MIME parameter "cps"
 in the "fmtp" attribute [7] is defined (see section 10).  It is used
 in Session Description Protocol (SDP) with the following syntax:
     a=fmtp:<format> cps=<integer>
 The <format> field is populated with the payload type that is used
 for text.  The <integer> field contains an integer representing the
 maximum number of characters that may be received per second.  The
 value shall be used as a mean value over any 10-second interval.  The
 default value is 30.
 In receipt of this parameter, devices MUST adhere to the request by
 transmitting characters at a rate at or below the specified <integer>
 value.  Examples of use in SDP are found in section 7.2.

Hellstrom & Jones Historic [Page 10] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

7. Examples

7.1. RTP Packetization Examples for the audio/t140c Format

 Below is an example of an audio/t140c RTP packet without redundancy.
  0                   1                   2                   3
  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |V=2|P|X| CC=0  |M|   T140c PT  |       sequence number         |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |                      timestamp (8000Hz)                       |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |           synchronization source (SSRC) identifier            |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |     T140block counter         | T.140 encoded data            |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               +---------------+
 |                                               |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 Below is an example of an RTP packet with one redundant T140block
 using audio/t140c payload format.  The primary data block is empty,
 which is the case when transmitting a packet for the sole purpose of
 forcing the redundant data to be transmitted in the absence of any
 new data.  Note that since this is the audio/t140c payload format,
 the redundant block of T.140 data is immediately preceded with a
 T140block counter.
  0                   1                   2                   3
  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |V=2|P|X| CC=0  |M|  "RED" PT   |   sequence number of primary  |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |               timestamp of primary encoding "P"               |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |           synchronization source (SSRC) identifier            |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |1|   T140c PT  |  timestamp offset of "R"  | "R" block length  |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |0|   T140c PT  |  "R" T140block counter        |               |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               +
 |               "R" T.140 encoded redundant data                |
 +                                               +---------------+
 |                                               |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Hellstrom & Jones Historic [Page 11] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

 As a follow-on to the previous example, the example below shows the
 next RTP packet in the sequence that does contain a new real
 T140block when using the audio/t140c payload format.  This example
 has 2 levels of redundancy and one primary data block.  Since the
 previous primary block was empty, no redundant data is included for
 that block.  This is because when using the audio/t140c payload
 format, any previously transmitted "empty" T140blocks are NOT
 included as redundant data in subsequent packets.
  0                   1                   2                   3
  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |V=2|P|X| CC=0  |M|  "RED" PT   |   sequence number of primary  |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |               timestamp of primary encoding "P"               |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |           synchronization source (SSRC) identifier            |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |1|   T140c PT  |  timestamp offset of "R1" | "R1" block length |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |0|   T140c PT  |  "R1" T140block counter       |               |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               +
 |               "R1" T.140 encoded redundant data               |
 +                                               +---------------+
 |                                               | "P" T140block |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | counter       |     "P" T.140 encoded primary data            |
 +-+-+-+-+-+-+-+-+                                               +
 |                                                               |
 +                                               +---------------+
 |                                               |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

7.2. SDP Examples

 Below is an example of SDP describing RTP text interleaved with G.711
 audio packets within the same RTP session from port 7200 and at a
 maximum text rate of 6 characters per second:
    m=audio 7200 RTP/AVP 0 98
    a=rtpmap:98 t140c/8000
    a=fmtp:98 cps=6
 Below is an example using RFC 2198 to provide the recommended two
 levels of redundancy to the text packets in an RTP session with
 interleaving text and G.711 at a text rate no faster than 20
 characters per second:

Hellstrom & Jones Historic [Page 12] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

    m=audio 7200 RTP/AVP 0 98 100
    a=rtpmap:98 t140c/8000
    a=fmtp:98 cps=20
    a=rtpmap:100 red/8000
    a=fmtp:100 98/98/98
 Note: While these examples utilize the RTP/AVP profile, it is not
 intended to limit the scope of this memo to use with only that
 profile.  Rather, any appropriate profile may be used in conjunction
 with this memo.

8. Security Considerations

 All of the security considerations from section 14 of RFC 3550 [2]
 apply.

8.1. Confidentiality

 Since the intention of the described payload format is to carry text
 in a text conversation, security measures in the form of encryption
 are of importance.  The amount of data in a text conversation session
 is low, and therefore any encryption method MAY be selected and
 applied to T.140 session contents or to the whole RTP packets.
 Secure Realtime Transport Protocol (SRTP) [13] provides a suitable
 method for ensuring confidentiality.

8.2. Integrity

 It may be desirable to protect the text contents of an RTP stream
 against manipulation.  SRTP [13] provides methods for providing
 integrity that MAY be applied.

8.3. Source Authentication

 Measures to make sure that the source of text is the intended one can
 be accomplished by a combination of methods.
 Text streams are usually used in a multimedia control environment.
 Security measures for authentication are available and SHOULD be
 applied in the registration and session establishment procedures, so
 that the identity of the sender of the text stream is reliably
 associated with the person or device setting up the session.  Once
 established, SRTP [13] mechanisms MAY be applied to ascertain that
 the source is maintained the same during the session.

Hellstrom & Jones Historic [Page 13] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

9. Congestion Considerations

 The congestion considerations from section 10 of RFC 3550 [2],
 section 6 of RFC 2198 [3], and any used profile (e.g., the part about
 congestion in section 2 of RFC 3551 [10]) apply with the following
 application-specific considerations.
 Automated systems MUST NOT use this format to send large amounts of
 text at a rate significantly above that which a human user could
 enter.
 Even if the network load from users of text conversation is usually
 very low, for best-effort networks an application MUST monitor the
 packet loss rate and take appropriate actions to reduce its sending
 rate if this application sends at higher rate than what TCP would
 achieve over the same path.  The reason is that this application, due
 to its recommended usage of two or more redundancy levels, is very
 robust against packet loss.  At the same time, due to the low bit-
 rate of text conversations, if one considers the discussion in RFC
 3714 [12], this application will experience very high packet loss
 rates before it needs to perform any reduction in the sending rate.
 If the application needs to reduce its sending rate, it SHOULD NOT
 reduce the number of redundancy levels below the default amount
 specified in section 4.  Instead, the following actions are
 RECOMMENDED in order of priority:
  1. 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.
  1. Limit the maximum rate of characters transmitted.
  1. 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.
  1. Exclude participants from the session.
 Please note that if the reduction in bit-rate achieved through the
 above measures is not sufficient, the only remaining action is to
 terminate the session.
 As guidance, some load figures are provided here as examples based on
 use of IPv4, including the load from IP, UDP, and RTP headers without
 compression.

Hellstrom & Jones Historic [Page 14] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

  1. Experience tells that a common mean character transmission rate

during a complete PSTN text telephony session in reality is around

   2 characters per second.
  1. A maximum performance of 20 characters per second is enough even

for voice-to-text applications.

  1. With the (unusually high) load of 20 characters per second, in a

language that make use of three-octet UTF-8 characters, two

   redundant levels, and 300 ms between transmissions, the maximum
   load of this application is 3500 bits/s.
  1. When the restrictions mentioned above are applied, limiting

transmission to 10 characters per second, using 5 s between

   transmissions, the maximum load of this application in a language
   that uses one octet per UTF-8 character is 300 bits/s.
 Note also, that this payload can be used in a congested situation as
 a last resort to maintain some contact when audio and video media
 need to be stopped.  The availability of one low bit-rate stream for
 text in such adverse situations may be crucial for maintaining some
 communication in a critical situation.

10. IANA Considerations

 This document defines one RTP payload format named "t140" and an
 associated MIME type "audio/t140c".  They have been registered by the
 IANA.

10.1. Registration of MIME Media Type audio/t140c

 MIME media type name: audio
 MIME subtype name: t140c
 Required parameters:
   rate: The RTP timestamp clock rate, which is equal to the
   sampling rate.  This parameter SHOULD have the same value as
   for any audio codec packets interleaved in the same RTP
   stream.
 Optional parameters:
   cps: The maximum number of characters that may be received
   per second.  The default value is 30.
 Encoding considerations: T.140 text can be transmitted with RTP
 as specified in RFC 4351.

Hellstrom & Jones Historic [Page 15] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

 Security considerations: See section 8 of RFC 4351.
 Interoperability considerations: None
 Published specification: ITU-T T.140 Recommendation.
                          RFC 4351.
 Applications which use this media type:
   Text communication systems and text conferencing tools that
   transmit text associated with audio and within the same RTP
   session as the audio, such as PSTN gateways that transmit
   audio and text signals between two PSTN textphone users
   over an IP network.
 Additional information:  This type is only defined for transfer
   via RTP.
   Magic number(s): None
   File extension(s): None
   Macintosh File Type Code(s): None
 Person & email address to contact for further information:
   Paul E. Jones
   E-mail: paulej@packetizer.com
 Intended usage: COMMON
 Author                        / Change controller:
   Paul E. Jones               | IETF avt WG delegated from the IESG
   paulej@packetizer.com       |

10.2. SDP Mapping of MIME Parameters

 The information carried in the MIME media type specification has a
 specific mapping to fields in the Session Description Protocol (SDP)
 [7], which is commonly used to describe RTP sessions.  When SDP is
 used to specify sessions employing the audio/t140c format, the
 mapping is as follows:
  1. The MIME type ("audio") goes in SDP "m=" as the media name.
  1. 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.
  1. The parameter "cps" goes in SDP "a=fmtp" attribute.

Hellstrom & Jones Historic [Page 16] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

  1. 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:
  1. The "cps" parameter is declarative. Both sides may provide a

value, which is independent of the other side.

11. Acknowledgements

 The authors want to thank Stephen Casner, Magnus Westerlund, and
 Colin Perkins for valuable support with reviews and advice on
 creation of this document; Mickey Nasiri at Ericsson Mobile
 Communication for providing the development environment; Michele
 Mizarro for verification of the usability of the payload format for
 its intended purpose; and Andreas Piirimets for editing support.

12. Normative References

 [1]  ITU-T Recommendation T.140 (1998) - Text conversation protocol
      for multimedia application, with amendment 1, (2000).
 [2]  Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
      "RTP: A Transport Protocol for Real-Time Applications", STD 64,
      RFC 3550, July 2003.
 [3]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M.,
      Bolot, J., Vega-Garcia, A., and S. Fosse-Parisis, "RTP Payload
      for Redundant Audio Data", RFC 2198, September 1997.
 [4]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
      Levels", BCP 14, RFC 2119, March 1997.
 [5]  ISO/IEC 10646-1: (1993), Universal Multiple Octet Coded
      Character Set.
 [6]  Yergeau, F., "UTF-8, a transformation format of ISO 10646", STD
      63, RFC 3629, November 2003.
 [7]  Handley, M. and V. Jacobson, "SDP: Session Description
      Protocol", RFC 2327, April 1998.

Hellstrom & Jones Historic [Page 17] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

 [8]  Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for
      Generic Forward Error Correction", RFC 2733, December 1999.
 [9]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
      Session Description Protocol (SDP)", RFC 3264, June 2002.
 [10] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
      Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
 [11] Postel, J., "Internet Protocol", STD 5, RFC 791, September 1981.

13. Informative References

 [12] Floyd, S. and J. Kempf, "IAB Concerns Regarding Congestion
      Control for Voice Traffic in the Internet", RFC 3714, March
      2004.
 [13] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
      Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
      3711, March 2004.
 [14] Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF Digits,
      Telephony Tones and Telephony Signals", RFC 2833, May 2000.
 [15] Hellstrom, G. and P. Jones, "RTP Payload for Text Conversation",
      RFC 4103, June 2005.
 [16] ITU-T Recommendation F.703, Multimedia Conversational Services,
      Nov 2000.
 [17] Casner, S. and P. Hoschka, "MIME Type Registration of RTP
      Payload Formats", RFC 3555, July 2003.

Hellstrom & Jones Historic [Page 18] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

Authors' Addresses

 Gunnar Hellstrom
 Omnitor AB
 Renathvagen 2
 SE-121 37 Johanneshov
 Sweden
 Phone: +46 708 204 288 / +46 8 556 002 03
 Fax:   +46 8 556 002 06
 EMail: gunnar.hellstrom@omnitor.se
 Paul E. Jones
 Cisco Systems, Inc.
 7025 Kit Creek Rd.
 Research Triangle Park, NC 27709
 USA
 Phone: +1 919 392 6948
 EMail: paulej@packetizer.com

Hellstrom & Jones Historic [Page 19] RFC 4351 RTP Payload for Text in an Audio Stream January 2006

Full Copyright Statement

 Copyright (C) The Internet Society (2006).
 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.
 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.

Intellectual Property

 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 procedures with respect to rights in RFC documents can be
 found in BCP 78 and BCP 79.
 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.

Acknowledgement

 Funding for the RFC Editor function is provided by the IETF
 Administrative Support Activity (IASA).

Hellstrom & Jones Historic [Page 20]

/data/webs/external/dokuwiki/data/pages/rfc/rfc4351.txt · Last modified: 2006/01/11 01:05 by 127.0.0.1

Donate Powered by PHP Valid HTML5 Valid CSS Driven by DokuWiki