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Framework for Real-Time Text over IP Using the Session Initiation Protocol (SIP)

The information below is for an old version of the document that is already published as an RFC.
Document Type
This is an older version of an Internet-Draft that was ultimately published as RFC 5194.
Authors Guido Gybels , Arnoud Wijk
Last updated 2015-10-14 (Latest revision 2008-04-04)
Replaces draft-vanwijk-sipping-toip
RFC stream Internet Engineering Task Force (IETF)
Intended RFC status Informational
Additional resources Mailing list discussion
Stream WG state (None)
Document shepherd (None)
IESG IESG state Became RFC 5194 (Informational)
Action Holders
Consensus boilerplate Unknown
Telechat date (None)
Responsible AD Jon Peterson
Send notices to (None)
SIPPING Workgroup                                    A. van Wijk, Editor
Internet Draft                                       G. Gybels, Editor
Category: Informational                              April 4, 2008
Expires: October 1, 2008

   Framework for real-time text over IP using the Session Initiation
                        Protocol (SIP)

Status of this Memo

  By submitting this Internet-Draft, each author represents that any
  applicable patent or other IPR claims of which he or she is aware have
  been or will be disclosed, and any of which he or she becomes aware
  will be disclosed, in accordance with Section 6 of BCP 79.

  Internet-Drafts are working documents of the Internet Engineering Task
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  This Internet-Draft will expire on October 1, 2008.

Copyright Notice

  Copyright (C) The IETF Trust (2008).


  This document lists the essential requirements for real-time Text-
  over-IP (ToIP) and defines a framework for implementation of all
  required functions based on the Session Initiation Protocol (SIP) and
  the Real-Time Transport Protocol (RTP). This includes interworking
  between Text-over-IP and existing text telephony on the PSTN and other

Table of Contents

   1. Introduction....................................................2
   2. Scope...........................................................3
   3. Terminology.....................................................4
   4. Definitions.....................................................4
   5. Requirements....................................................6
     5.1 General requirements for ToIP................................6

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     5.2 Detailed requirements for ToIP...............................7
       5.2.1 Session set-up and control requirements..................7
       5.2.2 Transport requirements...................................8
       5.2.3 Transcoding service requirements.........................9
       5.2.4 Presentation and User control requirements..............10
       5.2.5 Interworking requirements...............................11 PSTN Interworking requirements......................12 Cellular Interworking requirements..................12 Instant Messaging Interworking requirements.........12
   6. Implementation Framework.......................................13
     6.1 General implementation framework............................13
     6.2 Detailed implementation framework...........................13
       6.2.1 Session control and set-up..............................13 Pre-session set-up..................................13 Session Negotiations................................14
       6.2.2 Transport...............................................15
       6.2.3 Transcoding services....................................16
       6.2.4 Presentation and User control functions.................16 Progress and status information.....................16 Alerting............................................16 Text presentation...................................16 File storage........................................17
       6.2.5 Interworking functions..................................17 PSTN Interworking...................................18 Mobile Interworking.................................19
  Cellular "No-gain"..............................19
  Cellular Text Telephone Modem (CTM).............19
  Cellular "Baudot mode"..........................20
  Mobile data channel mode........................20
  Mobile ToIP.....................................20 Instant Messaging Interworking......................20 Multi-functional Combination gateways...............21 Character set transcoding...........................21
   7. Further recommendations for implementers and service providers.22
     7.1 Access to Emergency services................................22
     7.2 Home Gateways or Analog Terminal Adapters...................22
     7.3 User Mobility...............................................23
     7.4 Firewalls and NATs..........................................23
     7.5 Quality of Service..........................................23
   8. IANA Considerations............................................23
   9. Security Considerations........................................23
  10. Authors' Addresses.............................................24
  11. Contributors...................................................24
  12. References.....................................................24
    12.1 Normative references........................................24
    12.2 Informative references......................................26

1. Introduction

  For many years, real-time text has been in use as a medium for
  conversational, interactive dialogue between users in a similar way
  to how voice telephony is used. Such interactive text is different

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  from messaging and semi-interactive solutions like Instant Messaging 
  in that it offers an equivalent conversational experience to users
  who cannot, or do not wish to, use voice. It therefore meets a 
  different set of requirements from other text-based solutions already
  available on IP networks.

  Traditionally, deaf, hard of hearing and speech-impaired people are
  amongst the most prolific users of real-time, conversational, 
  text but, because of its interactivity, it is becoming popular amongst
  mainstream users as well. Real-time text conversation can be combined
  with other conversational media like video or voice.

  This document describes how existing IETF protocols can be used to
  implement a Text-over-IP solution (ToIP). This document describes
  therefore how to use a set of existing components and protocols and
  provides the requirements and rules for that resulting structure,
  which is why it is called a "framework", fitting commonly accepted
  dictionary definitions of that term.

  This ToIP framework is specifically designed to be compatible with 
  Voice-over-IP (VoIP), Video-over-IP and Multimedia-over-IP (MoIP) 
  environments. This ToIP framework also builds upon, and is compatible
  with, the high-level user requirements of deaf, hard of hearing and 
  speech-impaired users as described in RFC3351 [I]. It also meets 
  real-time text requirements of mainstream users.

  ToIP also offers an IP equivalent of analog text telephony services as
  used by deaf, hard of hearing, speech-impaired and mainstream users.

  The Session Initiation Protocol (SIP) [2] is the protocol of choice
  for control of Multimedia communications and Voice-over-IP (VoIP) in
  particular. It offers all the necessary control and signalling
  required for the ToIP framework.

  The Real-Time Transport Protocol (RTP) [3] is the protocol of choice
  for real-time data transmission, and its use for real-time text
  payloads is described in RFC4103 [4].

  This document defines a framework for ToIP to be used either by itself
  or as part of integrated, multi-media services, including Total
  Conversation [5].

2. Scope

  This document defines a framework for the implementation of real-time
  ToIP, either stand-alone or as a part of multimedia services,
  including Total Conversation [5]. It provides the:
  a. requirements for real-time text;
  b. requirements for ToIP interworking;
  c. description of ToIP implementation using SIP and RTP;
  d. description of ToIP interworking with other text services.

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3. Terminology

  The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
  "OPTIONAL" in this document are to be interpreted as described in
  RFC 2119 [6] and indicate requirement levels for compliant

4. Definitions

  Audio bridging: a function of an audio media bridge server, gateway or
  relay service that sends to each destination the combination of audio
  from all participants in a conference excluding the participant(s) at
  that destination. At the RTP level, this is an instance of the mixer
  function as defined in RFC 3550 [3].

  Cellular: a telecommunication network that has wireless access and can
  support voice and data services over very large geographical areas.
  Also called Mobile.

  Full duplex: media is sent independently in both directions.

  Half duplex: media can only be sent in one direction at a time or,
  if an attempt to send information in both directions is made, errors
  may be introduced into the presented media.

  Interactive text: another term for real-time text, as defined below.

  Real-time text: a term for real time transmission of text in a
  character-by-character fashion for use in conversational services,
  often as a text equivalent to voice based conversational services.
  Conversational text is defined in the ITU-T Framework for multimedia
  services, Recommendation F.700 [21].

  Text gateway: a function that transcodes between different forms of
  text transport methods, e.g., between ToIP in IP networks and Baudot
  or ITU-T V.21 text telephony in the PSTN.

  Textphone: also "text telephone". A terminal device that allows end-
  to-end real-time text communication using analog transmission. A
  variety of PSTN textphone protocols exists world-wide. A textphone can
  often be combined with a voice telephone, or include voice 
  communication functions for simultaneous or alternating use of text
  and voice in a call.

  Text bridging: a function of the text media bridge server, gateway
  (including transcoding gateways) or relay service analogous to that of
  audio bridging as defined above, except that text is the medium of

  Text relay service: a third-party or intermediary that enables
  communications between deaf, hard of hearing and speech-impaired

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  people and voice telephone users by translating between voice and
  real-time text in a call.

  Text telephony: analog textphone service.

  Total Conversation: a multimedia service offering real time
  conversation in video, real-time text and voice according to
  interoperable standards. All media streams flow in real time. (See
  ITU-T F.703 "Multimedia conversational services" [5].)

  Transcoding service: a service provided by a third-party User Agent
  that transcodes one stream into another. Transcoding can be done by
  human operators, in an automated manner, or by a combination of both
  methods. Within this document the term particularly applies to
  conversion between different types of media. A text relay service is
  an example of a transcoding service that converts between real-time
  text and audio.

  TTY: originally, an abbreviation for "teletype". Often used in North
  America as an alternative designation for a text telephone or
  textphone. Also called TDD, Telecommunication Device for the Deaf.

  Video relay service: a service that enables communications between
  deaf and hard of hearing people and hearing persons with voice
  telephones by translating between sign language and spoken language in
  a call.


  2G      Second generation cellular (mobile)
  2.5G    Enhanced second generation cellular (mobile)
  3G      Third generation cellular (mobile)
  ATA     Analog Telephone Adaptor
  CDMA    Code Division Multiple Access
  CLI     Calling Line Identification
  CTM     Cellular Text Telephone Modem
  ENUM    E.164 number storage in DNS (see RFC3761)
  GSM     Global System for Mobile Communications
  ISDN    Integrated Services Digital Network
  ITU-T   International Telecommunications Union-Telecommunications
          Standardisation Sector
  NAT     Network Address Translation
  PSTN    Public Switched Telephone Network
  RTP     Real Time Transport Protocol
  SDP     Session Description Protocol
  SIP     Session Initiation Protocol
  SRTP    Secure Real Time Transport Protocol
  TDD     Telecommunication Device for the Deaf
  TDMA    Time Division Multiple Access
  TTY     Analog textphone (Teletypewriter)
  ToIP    Real-time Text over Internet Protocol
  URI     Uniform Resource Identifier

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  UTF-8   UCS/Unicode Transformation Format-8
  VCO/HCO Voice Carry Over/Hearing Carry Over
  VoIP    Voice over Internet Protocol

5. Requirements

  The framework described in section 6 defines a real-time text-based
  conversational service that is the text equivalent of voice based
  telephony. This section describes the requirements that the framework
  is designed to meet and the functionality it should offer.

5.1 General requirements for ToIP

  Any framework for ToIP must be derived from the requirements of
  RFC3351 [I]. A basic requirement is that it must provide a
  standardized way for offering real-time text-based, conversational
  services that can be used as an equivalent to voice telephony by deaf,
  hard of hearing speech-impaired and mainstream users.

  It is important to understand that real-time text conversations are
  significantly different from other text-based communications like
  email or Instant Messaging. Real-time text conversations deliver an
  equivalent mode to voice conversations by providing transmission of
  text character by character as it is entered, so that the conversation
  can be followed closely and immediate interaction take place.

  Store-and-forward systems like email or messaging on mobile networks
  or non-streaming systems like instant messaging are unable to provide
  that functionality. In particular, they do not allow for smooth
  communication through a Text Relay Service.

  In order to make ToIP the text equivalent of voice services, ToIP
  needs to offer equivalent features in terms of conversationality to
  those provided by voice. To achieve that, ToIP needs to:

  a. offer real-time transport and presentation of the conversation;
  b. provide simultaneous transmission in both directions;
  c. support both point-to-point and multipoint communication;
  d. allow other media, like audio and video, to be used in conjunction
     with ToIP;
  e. ensure that the real-time text service is always available.

  Real-time text is a useful subset of Total Conversation as defined in
  ITU-T F.703 [5]. Total Conversation allows participants to use
  multiple modes of communication during the conversation, either at the
  same time or by switching between modes, e.g., between real-time text
  and audio.

  Deaf, hard-of-hearing and mainstream users may invoke ToIP services
  for many different reasons:

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  - because they are in a noisy environment, e.g., in a machine room of
    a factory where listening is difficult;
  - because they are busy with another call and want to participate in
    two calls at the same time;
  - for implementing text and/or speech recording services (e.g., text
    documentation/ audio recording) for legal purposes, for clarity or
    for flexibility;
  - to overcome language barriers through speech translation and/or
    transcoding services;
  - because of hearing loss, deafness or tinnitus as a result of the
    aging process or for any other reason, creating a need to replace or
    complement voice with real-time text in conversational sessions.

  In many of the above examples, real-time text may accompany speech.
  The text could be displayed side by side, or in a manner similar to 
  subtitling in broadcasting environments, or in any other suitable
  manner. This could occur with users who are hard of hearing and also
  for mixed media calls with both hearing and deaf people participating
  in the call.

  A ToIP user may wish to call another ToIP user, join a conference
  session involving several users, or initiate or join a multimedia
  session, such as a Total Conversation session.

  A common scenario for multipoint real-time text is conference calling
  with many participants. Implementers could for example use different
  colours to render different participants' text, or could create
  separate windows or rendering areas for each participant.

5.2 Detailed requirements for ToIP

  The following sections list individual requirements for ToIP. Each
  requirement has been given a unique identifier (R1, R2, etc). Section
  6 (Implementation Framework) describes how to implement ToIP based on
  these requirements and using existing protocols and techniques.

  The requirements are organized under the following headings:
  - session set-up and session control;
  - transport;
  - use of transcoding services;
  - presentation and user control;
  - interworking.

5.2.1 Session set-up and control requirements

  Conversations could be started using a mode other than real-time text.
  Simultaneous or alternating voice and real-time text is used by a
  large number of people who can send voice but must receive text (due
  to a hearing impairment), or who can hear but must send text (due to a
  speech impairment).

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  R1: It SHOULD be possible to start conversations in any mode (real-
  time text, voice, video) or combination of modes.

  R2: It MUST be possible for the users to switch to real-time text, or
  add real-time text as an additional modality, during the conversation.

  R3: Systems supporting ToIP MUST allow users to select any of the
  supported conversation modes at any time, including in mid-

  R4: Systems SHOULD allow the user to specify a preferred mode of
  communication in each direction, with the ability to fall back to
  alternatives that the user has indicated are acceptable.

  R5: If the user requests simultaneous use of real-time text and audio,
  and this is not possible because of constraints in the network, the
  system SHOULD try to establish text only communication if that is
  what the user has specified as his/her preference.

  R6: If the user has expressed a preference for real-time text,
  establishment of a connection including real-time text MUST have
  priority over other outcomes of the session setup.

  R7: It MUST be possible to use real-time text in conferences both as a
  medium of discussion between individual participants (for example, for
  sidebar discussions in real-time text while listening to the main
  conference audio) and for central support of the conference with
  real-time text interpretation of speech.

  R8: Session set up and negotiation of modalities MUST allow users to
  specify the language of the real-time text to be used. (It is
  RECOMMENDED that similar functionality be provided for the video part
  of the conversation, i.e. to specify the sign language being used).

  R9: Where certain session services are available for the audio media
  part of a session, these functions MUST also be supported for the
  real-time text media part of the same session. For example, call
  transfer must act on all media in the session.

5.2.2 Transport requirements

  ToIP will often be used to access a relay service [V], allowing real-
  time text users to communicate with voice users. With relay services,
  as well as in direct user-to-user conversation, it is crucial that
  text characters are sent as soon as possible after they are entered.
  While buffering may be done to improve efficiency, the delays SHOULD
  be kept minimal. In particular, buffering of whole lines of text will
  not meet character delay requirements.

  R10: Characters must be transmitted soon after entry of each character
  so that the maximum delay requirement can be met. An end-to-end delay
  time of one second is regarded as good, while users note and

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  appreciate shorter delays, down to 300ms. A delay of up to two seconds
  is possible to use.

  R11: Real-time text transmission from a terminal SHALL be performed
  character by character as entered, or in small groups of characters,
  so that no character is delayed from entry to transmission by more
  than 300 milliseconds.

  R12: It MUST be possible to transmit characters at a rate sufficient
  to support fast human typing as well as speech-to-text methods of
  generating real-time text. A rate of 30 characters per second is
  regarded as sufficient.

  R13: A ToIP service MUST be able to deal with international character

  R14: Where it is possible, loss or corruption of real-time text during
  transport SHOULD be detected and the user should be informed.

  R15: Transport of real-time text SHOULD be as robust as possible, so
  as to minimize loss of characters.

  R16: It SHOULD be possible to send and receive real-time text

5.2.3 Transcoding service requirements

  If the User Agents of different participants indicate that there is an
  incompatibility between their capabilities to support certain media
  types, e.g. one User Agent only offering T.140 over IP as described in
  RFC4103 [4] and the other one only supporting audio, the user might
  want to invoke a transcoding service.

  Some users may indicate their preferred modality to be audio while
  others may indicate real-time text. In this case, transcoding services
  might be needed for text-to-speech (TTS) and speech-to-text (STT).
  Other examples of possible scenarios for including a relay service in
  the conversation are: text bridging after conversion from speech,
  audio bridging after conversion from real-time text, etc.

  A number of requirements, motivations and implementation guidelines
  for relay service invocation can be found in RFC 3351 [I].

  R17: It MUST be possible for users to invoke a transcoding service
  where such service is available.

  R18: It MUST be possible for users to indicate their preferred
  modality (e.g. ToIP).

  R19: It MUST be possible to negotiate the requirements for transcoding
  services in real time in the process of setting up a call.

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  R20: It MUST be possible to negotiate the requirements for transcoding
  services in mid-call, for the immediate addition of those services to
  the call.

  R21: Communication between the end participants SHOULD continue after
  the addition or removal of a text relay service, and the effect of the
  change should be limited in the users' perception to the direct effect
  of having or not having the transcoding service in the connection.

  R22: When setting up a session, it MUST be possible for a user to
  specify the type of relay service requested (e.g., speech to text or
  text to speech). The specification of a type of relay SHOULD include
  a language specifier.

  R23: It SHOULD be possible to route the session to a preferred relay
  service even if the user invokes the session from another region or
  network than that usually used.

  R24: It is RECOMMENDED that ToIP implementations make the invocation
  and use of relay services as easy as possible.

5.2.4 Presentation and User control requirements

  A user should never be in doubt about the status of the session, even
  if the user is unable to make use of the audio or visual indication.
  For example, tactile indications could be used by deafblind

  R25: User Agents for ToIP services MUST have alerting methods (e.g.,
  for incoming sessions) that can be used by deaf and hard of hearing
  people or provide a range of alternative, but equivalent, alerting
  methods that can be selected by all users, regardless of their

  R26: Where real-time text is used in conjunction with other media,
  exposure of user control functions through the User Interface needs to
  be done in an equivalent manner for all supported media. For example,
  it must be possible for the user to select between audio, visual or
  tactile prompts, or all must be supplied.

  R27: If available, identification of the originating party (for
  example in the form of a URI or a CLI) MUST be clearly presented to
  the user in a form suitable for the user BEFORE the session invitation
  is answered.

  R28: When a session invitation involving ToIP originates from a PSTN
  text telephone (e.g. transcoded via a text gateway), this SHOULD be
  indicated to the user. The ToIP client MAY adjust the presentation of
  the real-time text to the user as a consequence.

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  R29: An indication SHOULD be given to the user when real-time text is
  available during the call, even if it is not invoked at call setup
  (e.g. when only voice and/or video is used initially).

  R30: The user MUST be informed of any change in modalities.

  R31: Users MUST be presented with appropriate session progress
  information at all times.

  R32: Systems for ToIP SHOULD support an answering machine function,
  equivalent to answering machines on telephony networks.

  R33: If an answering machine function is supported, it MUST support at
  least 160 characters for the greeting message. It MUST support
  incoming text message storage of a minimum of 4096 characters,
  although systems MAY support much larger storage. It is RECOMMENDED
  that systems support storage of at least 20 incoming messages of up to
  16000 characters per message.

  R34: When the answering machine is activated, user alerting SHOULD
  still take place. The user SHOULD be allowed to monitor the auto-
  answer progress and where this is provided the user SHOULD be allowed
  to intervene during any stage of the answering machine procedure and
  take control of the session.

  R35: It SHOULD be possible to save the text portion of a conversation.

  R36: The presentation of the conversation SHOULD be done in such a way
  that users can easily identify which party generated any given portion
  of text.

  R37: ToIP SHOULD handle characters such as new line, erasure and
  alerting during a session as specified in ITU-T T.140 [8].

5.2.5 Interworking requirements

  There is a range of existing real-time text services. There is also a
  range of network technologies that could support real-time text

  Real-time/interactive texting facilities exist already in various
  forms and on various networks. In the PSTN, they are commonly referred
  to as text telephony.

  Text gateways are used for converting between different protocols for
  text conversation. They can be used between networks or within
  networks where different transport technologies are used.

  R38: ToIP SHOULD provide interoperability with text conversation
  features in other networks, for instance the PSTN.

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  R39: When communicating via a gateway to other networks and protocols,
  the ToIP service SHOULD support the functionality for alternating or
  simultaneous use of modalities as offered by the interworking network.

  R40: Calling party identification information, such as CLI, MUST be
  passed by gateways and converted to an appropriate form if required.

  R41: When interworking with other networks and services, the ToIP
  service SHOULD provide buffering mechanisms to deal with delays in
  call setup, differences in transmission speeds and/or to interwork
  with half duplex services. PSTN Interworking requirements

  Analog text telephony is used in many countries, mainly by deaf, hard
  of hearing and speech-impaired individuals.

  R42: ToIP services MUST provide interworking with PSTN legacy text
  telephony devices.

  R43: When interworking with PSTN legacy text telephony services,
  alternating text and voice function MAY be supported. (Called "voice
  carry over (VCO) and hearing carry over (HCO)"). Cellular Interworking requirements

  As mobile communications have been adopted widely, various solutions
  for real-time texting while on the move were developed. ToIP services
  should provide interworking with such services as well.

  Alternative means of transferring the Text telephony data have been
  developed when TTY services over cellular were mandated by the FCC in
  the USA. They are the a) "No-gain" codec solution, and b) the Cellular
  Text Telephony Modem (CTM) solution [7] both collectively called
  "Baudot mode" solution in the USA.

  The GSM and 3G standards from 3GPP make use of the CTM modem in the
  voice channel for text telephony. However, implementations also exist
  that use the data channel to provide such functionality. Interworking
  with these solutions should be done using text gateways that set up
  the data channel connection at the GSM side and provide ToIP at the
  other side.

  R44: a ToIP service SHOULD provide interworking with mobile text
  conversation services. Instant Messaging Interworking requirements

  Many people use Instant Messaging to communicate via the Internet
  using text. Instant Messaging usually transfers blocks of text rather
  than streaming as is used by ToIP. Usually a specific action is
  required by the user to activate transmission, such as pressing the

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  ENTER key or a send button. As such, it is not a replacement for ToIP
  and in particular does not meet the needs for real time conversations
  including those of deaf, hard of hearing and speech-impaired users as
  defined in RFC 3351 [I]. It is less suitable for communications
  through a relay service [V].

  The streaming nature of ToIP provides a more direct conversational
  user experience and, when given the choice, users may prefer ToIP.

  R45: a ToIP service MAY provide interworking with Instant Messaging

6. Implementation Framework

  This section describes an implementation framework for ToIP that meets
  the requirements and offers the functionality as set out in section 5.
  The framework presented here uses existing standards that are already
  commonly used for voice based conversational services on IP networks.

6.1 General implementation framework

  This framework specifies the use of the Session Initiation Protocol
  (SIP) [2] to set up, control and tear down the connections between
  ToIP users whilst the media is transported using the Real-Time
  Transport Protocol (RTP) [3] as described in RFC 4103 [4].

  RFC 4504 describes how to implement support for real-time text in SIP
  telephony devices [II].

6.2 Detailed implementation framework

6.2.1 Session control and set-up

  ToIP services MUST use the Session Initiation Protocol (SIP) [2] for
  setting up, controlling and terminating sessions for real-time text
  conversation with one or more participants and possibly including
  other media like video or audio. The Session Description Protocol
  (SDP) used in SIP to describe the session is used to express the
  attributes of the session and to negotiate a set of compatible media

  SIP [2] allows participants to negotiate all media including real-time
  text conversation [4]. ToIP services can provide the ability to set up
  conversation sessions from any location as well as provision for
  privacy and security through the application of standard SIP
  techniques. Pre-session set-up

  The requirements of the user to be reached at a consistent address and
  to store preferences for evaluation at session setup are met by pre-
  session setup actions. That includes storing of registration

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  information in the SIP registrar, to provide information about how a
  user can be contacted. This will allow sessions to be set up rapidly
  and with proper routing and addressing.

  The need to use real-time text as a medium of communications can be
  expressed by users during registration time. Two situations need to be
  considered in the pre-session setup environment:

  a. User Preferences: It MUST be possible for a user to indicate a
     preference for real-time text by registering that preference with a
     SIP server that is part of the ToIP service.

  b. Server support of User Preferences: SIP servers that support ToIP
     services MUST have the capability to act on calling user
     preferences for real-time text in order to accept or reject the
     session. The actions taken can be based on the called users
     preferences defined as part of the pre-session setup registration.
     For example, if the user is called by another party, and it is
     determined that a transcoding server is needed, the session should
     be re-directed or otherwise handled accordingly.

  The ability to include a transcoding service MUST NOT require user
  registration in any specific SIP registrar, but MAY require
  authorisation of the SIP registrar to invoke the service.

  A point-to-point session takes place between two parties. For ToIP,
  one or both of the communicating parties will indicate real-time text
  as a possible or preferred medium for conversation using SIP in the
  session setup.

  The following features MAY be implemented to facilitate the session
  establishment using ToIP:

  a. Caller Preferences: SIP headers (e.g., Contact) [10] can be used to
     show that real-time text is the medium of choice for

  b. Called Party Preferences [11]: The called party being passive can
     formulate a clear rule indicating how a session should be handled
     either using real-time text as a preferred medium or not, and
     whether a designated SIP proxy needs to handle this session or it
     will be handled in the SIP User Agent.

  c. SIP Server support for User Preferences: It is RECOMMENDED that SIP
     servers also handle the incoming sessions in accordance with
     preferences expressed for real-time text. The SIP Server can also
     enforce ToIP policy rules for communications (e.g. use of the
     transcoding server for ToIP). Session Negotiations

  The Session Description Protocol (SDP) used in SIP [2] provides the

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  capabilities to indicate real-time text as a medium in the session
  setup. RFC 4103 [4] uses the RTP payload types "text/red" and
  "text/t140" for support of ToIP which can be indicated in the SDP as a
  part of the SIP INVITE, OK and SIP/200/ACK media negotiations. In
  addition, SIPs offer/answer model [12] can also be used in conjunction
  with other capabilities including the use of a transcoding server for
  enhanced session negotiations [III,IV,13].

6.2.2 Transport

  ToIP services MUST support the Real-Time Transport Protocol (RTP) [3]
  according to the specification of RFC 4103 [3] for the transport of
  real-time text between participants.

  RFC 4103 describes the transmission of T.140 [8] real-time text on IP

  In order to enable the use of international character sets, the
  transmission format for real-time text conversation SHALL be UTF-8
  [14], in accordance with ITU-T T.140.

  If real-time text is detected to be missing after transmission, there
  SHOULD be a "text loss" indication in the real-time text as specified
  in T.140 Addendum 1 [8].

  The redundancy method of RFC 4103 [4] SHOULD be used to significantly
  increase the reliability of the real-time text transmission. A
  redundancy level using 2 generations gives very reliable results and
  is therefore strongly RECOMMENDED.

  In order to avoid exceeding the capabilities of sender, receiver or
  network (congestion), the transmission rate SHOULD be kept at or
  below 30 characters per second, which is the default maximum rate as
  specified in RFC 4103 [4]. Lower rates MAY be negotiated when needed
  through the "cps" parameter as specified in RFC 4103 [4].

  Real-time text capability is announced in SDP by a declaration similar
  to this example:

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

  By having this single coding and transmission scheme for real-time
  text defined in the SIP session control environment, the opportunity
  for interoperability is optimized. However, if good reasons exist,
  other transport mechanisms MAY be offered and used for the T.140 coded
  text provided that proper negotiation is introduced, but RFC 4103 [4]
  transport MUST be used as both the default and the fallback transport.

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6.2.3 Transcoding services

  Invocation of a transcoding service MAY happen automatically when the
  session is being set up based on any valid indication or negotiation
  of supported or preferred media types. A transcoding framework
  document using SIP [III] describes invoking relay services, where the
  relay acts as a conference bridge or uses the third party control
  mechanism. ToIP implementations SHOULD support this transcoding

6.2.4 Presentation and User control functions Progress and status information

  Session progress information SHOULD use simple language so that as
  many users as possible can understand it. The use of jargon or
  ambiguous terminology SHOULD be avoided. It is RECOMMENDED that text
  information be used together with icons to symbolise the session
  progress information.

  In summary, it SHOULD be possible to observe indicators about:
  - Incoming session
  - Availability of real-time text, voice and video channels
  - Session progress
  - Incoming real-time text
  - Any loss in incoming real-time text
  - Typed and transmitted real-time text. Alerting

  For users who cannot use the audible alerter for incoming sessions, it
  is RECOMMENDED to include a tactile as well as a visual indicator.

  Among the alerting options are alerting by the User Agent's User
  Interface and specific alerting User Agents registered to the same
  registrar as the main User Agent.

  It should be noted that external alerting systems exist and one common
  interface for triggering the alerting action is a contact closure
  between two conductors. Text presentation

  Requirement R32 states that, in the display of text conversations,
  users must be able to distinguish easily between different speakers.
  This could be done using color, positioning of the text (i.e. incoming
  real-time text and outgoing real-time text in different display
  areas), by in-band identifiers of the parties or by a combination of
  any of these techniques.

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  Requirement R31 recommends that ToIP systems allow the user to save
  text conversations. This SHOULD be done using a standard file format.
  For example: a UTF-8 text file in XHTML format [15] including
  timestamps, party names (or addresses) and the conversation text.

6.2.5 Interworking functions

  A number of systems for real-time text conversation already exist as
  well as a number of message oriented text communication systems.
  Interoperability is of interest between ToIP and some of these

  Interoperation of half-duplex and full-duplex protocols, and between
  protocols that have different data rates, may require text buffering.
  Some intelligence will be needed to determine when to change direction
  when operating in half-duplex mode. Identification may be required of
  half-duplex operation either at the "user" level (ie. users must
  inform each other) or at the "protocol" level (where an indication
  must be sent back to the Gateway). However, special care needs to be
  taken to provide the best possible real-time performance.

  Buffering schemes SHOULD be dimensioned to adjust for receiving at 30
  characters per second and transmitting at 6 characters per second for
  up to 4 minutes (i.e. less than 3000 characters).

  When converting between simultaneous voice and text on the IP side,
  and alternating voice and text on the other side of a gateway, a
  conflict can occur if the IP user transmits both audio and text at the
  same time. In such situations, text transmission SHOULD have
  precedence, so that while text is transmitted, audio is lost.

  Transcoding of text to and from other coding formats may need to take
  place in gateways between ToIP and other forms of text conversation,
  for example to connect to a PSTN text telephone.

  Session set-up through gateways to other networks may require the use
  of specially formatted addresses or other mechanisms for invoking
  those gateways.

  ToIP interworking requires a method to invoke a text gateway. These
  text gateways act as User Agents at the IP side. The capabilities of
  the gateway during the call will be determined by the call
  capabilities of the terminal that is using the gateway. For example, a
  PSTN textphone is generally only able to receive voice and real-time
  text, so the gateway will only allow ToIP and audio.

  Examples of possible scenarios for invocation of the text gateway are:

  a. PSTN textphone users dial a prefix number before dialing out.

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  b. Separate real-time text subscriptions, linked to the phone number
     or terminal identifier/ IP address.
  c. Real-time text capability indicators.
  d. Real-time text preference indicators.
  e. Listen for V.18 modem modulation text activity in all PSTN calls
     and routing of the call to an appropriate gateway.
  f. Call transfer request by the called user.
  g. Placing a call via the web, and using one of the methods described
  h. A text gateway with its own telephone number and/or SIP address.
     (This requires user interaction with the gateway to place a call).
  i. ENUM address analysis and number plan.
  j. Number or address analysis leads to a gateway for all PSTN calls. PSTN Interworking

  Analog text telephony is cumbersome because of incompatible national
  implementations where interworking was never considered. A large
  number of these implementations have been documented in ITU-T V.18
  [16], which also defines the modem detection sequences for the
  different text protocols. The modem type identification may in rare
  cases take considerable time depending on user actions.

  To resolve analog textphone incompatibilities, text telephone gateways
  are needed to transcode incoming analog signals into T.140 and vice
  versa. The modem capability exchange time can be reduced by the text
  telephone gateways initially assuming the analog text telephone
  protocol used in the region where the gateway is located. For example,
  in the USA, Baudot [VI] might be tried as the initial protocol. If
  negotiation for Baudot fails, the full V.18 modem capability exchange
  will take place. In the UK, ITU-T V.21 [VII] might be the first

  In particular transmission of real-time text on PSTN networks takes
  place using a variety of codings and modulations, including ITU-T 
  V.21 [VII], Baudot [VI], DTMF, V.23 [VIII] and others. Many
  difficulties have arisen as a result of this variety in text
  telephony protocols and the ITU-T V.18 [16] standard was developed to
  address some of these issues.

  ITU-T V.18 [16] offers a native text telephony method plus it defines
  interworking with current protocols. In the interworking mode, it will
  recognise one of the older protocols and fall back to that
  transmission method when required.

  Text gateways MUST use the ITU-T V.18 [16] standard at the PSTN side.
  A text gateway MUST act as a SIP User Agent on the IP side and support
  RFC 4103 real-time text transport.

  While ToIP allows receiving and sending real-time text simultaneously
  and is displayed on a split screen, many analog text telephones
  require users to take turns typing. This is because many text

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  telephones operate strictly half duplex. Only one can transmit text at
  a time. The users apply strict turn-taking rules.

  There are several text telephones which communicate in full duplex,
  but merge transmitted text and received text in the same line in the
  same display window. Here too the users apply strict turn taking

  Native V.18 text telephones support full duplex and separate display
  from reception and transmission so that the full duplex capability
  can be used fully. Such devices could use the ToIP split screen as
  well, but almost all text telephones use a restricted character set
  and many use low text transmission speeds (4 to 7 characters per

  That is why it is important for the ToIP user to know that he or she
  is connected with an analog text telephone. The session description
  [9] SHOULD contain an indication that the other endpoint for the call
  is a PSTN textphone (e.g. connected via an ATA or through a text 
  gateway). This means that the textphone user may be used to formal
  turn taking during the call. Mobile Interworking

  Mobile wireless (or Cellular) circuit switched connections provide a
  digital real-time transport service for voice or data. The access
  technologies include GSM, CDMA, TDMA, iDen and various 3G
  technologies as well as WiFi or WiMAX.

  ToIP may be supported over the cellular wireless packet switched
  service. It interfaces to the Internet.

  The following sections describe how mobile text telephony is
  supported. Cellular "No-gain"

  The "No-gain" text telephone transporting technology uses specially
  modified EFR [17] and EVR [18] speech vocoders in mobile terminals
  used to provide a text telephony call. It provides full duplex
  operation and supports alternating voice and text ("VCO/HCO"). It is
  dedicated to CDMA and TDMA mobile technologies and the US Baudot (i.e.
  45 bit/s) type of text telephones. Cellular Text Telephone Modem (CTM)

  CTM [7] is a technology independent modem technology that provides the
  transport of text telephone characters at up to 10 characters/sec
  using modem signals that can be carried by many voice codecs and uses
  a highly redundant encoding technique to overcome the fading and cell
  changing losses.

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  This term is often used by cellular terminal suppliers for a cellular
  phone mode that allows TTYs to operate into a cellular phone and to
  communicate with a fixed line TTY. Thus it is a common name for the
  "No-Gain" and the CTM solutions when applied to the Baudot type
  textphones. Mobile data channel mode

  Many mobile terminals allow the use of the circuit switched data
  channel to transfer data in real-time. Data rates of 9600 bit/s are
  usually supported on the 2G mobile network. Gateways provide
  interoperability with PSTN textphones. Mobile ToIP

  ToIP could be supported over mobile wireless packet switched services
  that interface to the Internet. For 3GPP 3G services, ToIP support is
  described in 3G TS 26.235 [19]. Instant Messaging Interworking

  Text gateways MAY be used to allow interworking between Instant
  Messaging systems and ToIP solutions. Because Instant Messaging is
  based on blocks of text, rather than on a continuous stream of
  characters like ToIP, gateways MUST transcode between the two formats.
  Text gateways for interworking between Instant Messaging and ToIP MUST
  apply a procedure for bridging the different conversational formats of
  real-time text versus text messaging. The following advice may improve
  user experience for both parties in a call through a messaging

  a. Concatenate individual characters originating at the ToIP side into
     blocks of text.

  b. When the length of the concatenated message becomes longer than 50
     characters, the buffered text SHOULD be transmitted to the Instant
     Messaging side as soon as any non-alphanumerical character is
     received from the ToIP side.

  c. When a new line indicator is received from the ToIP side, the
     buffered characters up to that point, including the carriage return
     and/or line feed characters, SHOULD be transmitted to the Instant
     Messaging side.

  d. When the ToIP side has been idle for at least 5 seconds, all
     buffered text up to that point SHOULD be transmitted to the Instant
     Messaging side.

  e. Text Gateways must be capable to maintain the real-time performance
     for ToIP while providing the interworking services.

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  It is RECOMMENDED that during the session, both users be constantly
  updated on the progress of the text input. Many Instant Messaging
  protocols signal that a user is typing to the other party in the
  conversation. Text gateways between such Instant Messaging protocols
  and ToIP MUST provide this signalling to the Instant Messaging side
  when characters start being received, or at the beginning of the

  At the ToIP side, an indicator of writing the Instant Message MUST be
  present where the Instant Messaging protocol provides one. For
  example, the real-time text user MAY see ". . . waiting for replying
  IM. . . " and when 5 seconds have passed another . (dot) can be shown.

  Those solutions will reduce the difficulties between streaming and
  blocked text services.

  Even though the text gateway can connect Instant Messaging and ToIP,
  the best solution is to take advantage of the fact that the user
  interfaces and the user communities for instant messaging and ToIP
  telephony are very similar. After all, the character input, the
  character display, Internet connectivity and SIP stack can be the same
  for Instant Messaging (SIMPLE) and ToIP. Thus, the user may simply use
  different applications for ToIP and text messaging in the same

  Devices that implement Instant Messaging SHOULD implement ToIP as
  described in this document so that a more complete text communication
  service can be provided. Multi-functional Combination gateways

  In practice many interworking gateways will be implemented as gateways
  that combine different functions. As such, a text gateway could be
  built to have modems to interwork with the PSTN and support both
  Instant Messaging as well as ToIP. Such interworking functions are
  called Combination gateways.

  Combination gateways could provide interworking between all of their
  supported text based functions. For example, a Text gateway that has
  modems to interwork with the PSTN and that support both Instant
  Messaging and ToIP could support the following interworking functions:

  - PSTN text telephony to ToIP.
  - PSTN text telephony to Instant Messaging.
  - Instant Messaging to ToIP. Character set transcoding

  Gateways between the ToIP network and other networks MAY need to
  transcode text streams. ToIP makes use of the ISO 10646 character set.
  Most PSTN textphones use a 7-bit character set, or a character set
  that is converted to a 7-bit character set by the V.18 modem.

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  When transcoding between character sets and T.140 in gateways, special
  consideration MUST be given to the national variants of the 7-bit
  codes, with national characters mapping into different codes in the
  ISO 10646 code space. The national variant to be used could be
  selectable by the user on a per call basis, or be configured as a
  national default for the gateway.

  The indicator of missing text in T.140, specified in T.140 amendment
  1, cannot be represented in the 7-bit character codes. Therefore the
  indicator of missing text SHOULD be transcoded to the ' (apostrophe)
  character in legacy text telephone systems, where this character
  exists. For legacy systems where the ' character does not exist, the .
  (full stop) character SHOULD be used instead.

7. Further recommendations for implementers and service providers

7.1 Access to Emergency services

  It must be possible to place an emergency call using ToIP and it must
  be possible to use a relay service in such call. The emergency service
  provided to users utilising the real-time text medium must be
  equivalent to the emergency service provided to users utilising speech
  or other media.

  A text gateway must be able to route real-time text calls to emergency
  service providers when any of the recognised emergency numbers that
  support text communications for the country or region are called e.g.
  "911" in USA and "112" in Europe. Routing real-time text calls to
  emergency services may require the use of a transcoding service.

  A text gateway with cellular wireless packet switched services must be
  able to route real-time text calls to emergency service providers when
  any of the recognized emergency numbers that support real-time text
  communication for the country is called.

7.2 Home Gateways or Analog Terminal Adapters

  Analog terminal adapters (ATA) using SIP based IP communication and
  RJ-11 connectors for connecting traditional PSTN devices SHOULD enable
  connection of legacy PSTN text telephones [II].

  These adapters SHOULD contain V.18 modem functionality, voice handling
  functionality, and conversion functions to/from SIP based ToIP with
  T.140 transported according to RFC 4103 [3], in a similar way as it
  provides interoperability for voice sessions.

  If a session is set up and text/t140 capability is not declared by the
  destination endpoint (by the end-point terminal or the text gateway in
  the network at the end-point), a method for invoking a transcoding
  server SHALL be used. If no such server is available, the signals from
  the textphone MAY be transmitted in the voice channel as audio with
  high quality of service.

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  NOTE: It is preferred that such analog terminal adaptors do use RFC
  4103 [4] on board and thus act as a text gateway. Sending textphone
  signals over the voice channel is undesirable due to possible
  filtering and compression and packet loss between the end-points. This
  can result in character loss in the textphone conversation or even not
  allowing the textphones to connect to each other.

7.3 User Mobility

  ToIP User Agents SHOULD use the same mechanisms as other SIP User
  Agents to resolve mobility issues. It is RECOMMENDED that users use a
  SIP address, resolved by a SIP registrar, to enable basic user
  mobility. Further mechanisms are defined for all session types for 3G
  IP multimedia systems.

7.4 Firewalls and NATs

  ToIP uses the same signalling and transport protocols as VoIP. Hence,
  the same firewall and NAT solutions and network functionality that
  apply to VoIP MUST also apply to ToIP.

7.5 Quality of Service

  Where Quality of Service (QoS) mechanisms are used, the real-time text
  streams should be assigned appropriate QoS characteristics, so that
  the performance requirements can be met and the real-time text stream
  is not degraded unfavourably in comparison to voice performance in
  congested situations.

8. IANA Considerations

  There are no IANA considerations for this specification.

9. Security Considerations

  User confidentiality and privacy need to be met as described in SIP
  [2]. For example, nothing should reveal in an obvious way the fact 
  that the ToIP user might be a person with a hearing or speech
  impairment. It is up to the ToIP user to make his or her hearing or
  speech impairment public. If a transcoding server is being used, 
  this SHOULD be as transparent as possible. However, it might still be
  possible to discern that a user might be hearing or speech impaired
  based on the attributes present in SDP, although the intention is
  that mainstream users might also choose to use ToIP.
  Encryption SHOULD be used on end-to-end or hop-by-hop basis as
  described in SIP [2] and SRTP [20].

  Authentication MUST be provided for users in addition to message
  integrity and access control.

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  Protection against Denial-of-service (DoS) attacks needs to be
  provided considering the case that the ToIP users might need
  transcoding servers.

10. Authors' Addresses

    Guido Gybels
    Department of New Technologies
    RNID, 19-23 Featherstone Street
    London EC1Y 8SL, UK
    Tel +44-20-7294 3713
    Txt +44-20-7296 8001 Ext 3713
    Fax +44-20-7296 8069
    Arnoud A. T. van Wijk
    Real-Time Text Taskforce (R3TF)

11. Contributors

  The following people contributed to this document: Willem Dijkstra,
  Barry Dingle, Gunnar Hellstrom, Radhika R. Roy, Henry Sinnreich and
  Gregg C Vanderheiden.

  The content and concepts within are a product of the SIPPING Working
  Group. Tom Taylor (Nortel) acted as independent reviewer and
  contributed significantly to the structure and content of this

12. References

12.1 Normative references

   1. S. Bradner, "Intellectual Property Rights in IETF Technology", 
      BCP 79, RFC 3979, IETF, March 2005.

   2. J. Rosenberg, H. Schulzrinne, G. Camarillo, A. R. Johnston, J.
      Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: Session
      Initiation Protocol", RFC 3621, IETF, June 2002.

   3. H. Schulzrinne, S.Casner, R. Frederick, V. Jacobsone, "RTP: A
      Transport Protocol for Real-Time Applications", RFC 3550, IETF,
      July 2003.

   4. G. Hellstrom, P. Jones, "RTP Payload for Text Conversation",
      RFC 4103, IETF, June 2005.

   5. ITU-T Recommendation F.703,"Multimedia Conversational Services",
      November 2000.

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   6. S. Bradner, "Key words for use in RFCs to Indicate Requirement
      Levels", BCP 14, RFC 2119, IETF, March 1997

   7. 3GPP TS 26.226  "Cellular Text Telephone Modem Description" (CTM).

   8. ITU-T Recommendation T.140, "Protocol for Multimedia Application
      Text Conversation" (February 1998) and Addendum 1 (February 2000).

   9. M. Handley, V. Jacobson, C. Perkins, "SDP: Session Description
      Protocol", RFC 4566, IETF, July 2006. 

  10. J. Rosenberg, H. Schulzrinne, P. Kyzivat, "Indicating User Agent
      Capabilities in the Session Initiation Protocol (SIP)", RFC 3840,
      IETF, August 2004

  11. J. Rosenberg, H. Schulzrinne, P. Kyzivat, "Caller Preferences for
      the Session Initiation Protocol (SIP)", RFC 3841, IETF,
      August 2004

  12. J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with the
      Session Description Protocol (SDP)", RFC 3624, IETF, June 2002.

  13. G. Camarillo, H. Schulzrinne, E. Burger, and A. van Wijk,
      "Transcoding Services Invocation in the Session Initiation
      Protocol (SIP) Using Third Party Call Control (3pcc)" RFC 4117,
      IETF, June 2005.

  14. Yergeau, F., "UTF-8, a transformation format of ISO 10646",
      RFC 3629, IETF,November 2003.

  15. "XHTML 1.0: The Extensible HyperText Markup Language: A
      Reformulation of HTML 4 in XML 1.0", W3C Recommendation. Available

  16. ITU-T Recommendation V.18,"Operational and Interworking
      Requirements for DCEs operating in Text Telephone Mode",
      November 2000.

  17. TIA/EIA/IS-823-A  "TTY/TDD Extension to TIA/EIA-136-410 Enhanced
      Full Rate Speech Codec (must used in conjunction with

  18. TIA/EIA/IS-127-2 "Enhanced Variable Rate Codec, Speech Service
      Option 3 for Wideband Spread Spectrum Digital Systems. 
      Addendum 2."

  19. "IP Multimedia default codecs". 3GPP TS 26.235

  20. Baugher, McGrew, Carrara, Naslund, Norrman, "The Secure Real Time
      Transport Protocol (SRTP)", RFC 3711, IETF, March 2004.

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  21. ITU-T Recommendation F.700,"Framework Recommendation for
      Multimedia Services", November 2000.

12.2 Informative references

  I.   Charlton, Gasson, Gybels, Spanner, van Wijk, "User Requirements
       for the Session Initiation Protocol (SIP) in Support of Deaf, 
       Hard of Hearing and Speech-impaired Individuals", RFC 3351,
       IETF, August 2002.

  II.  H. Sinnreich, S. Lass,  and C. Stredicke, "SIP Telephony Device
       Requirements and Configuration" RFC 4504, IETF, May 2006.

  III. G. Camarillo, "Framework for Transcoding with the Session
       Initiation Protocol", IETF, May 2006 -  Work in progress.

  IV.  G. Camarillo, "The SIP Conference Bridge Transcoding Model",
       IETF, January 2006 - Work in Progress.

  V.   European Telecommunications Standards Institute (ETSI), "Human
       Factors (HF); Guidelines for Telecommunication Relay Services for
       Text Telephones". TR 101 806, June 2000.

  VI.  TIA/EIA/825 "A Frequency Shift Keyed Modem for Use on the Public
       Switched Telephone Network." (The specification for 45.45 and 50
       bit/s TTY modems.)

  VII. International Telecommunication Union (ITU), "300 bits per second
       duplex modem standardized for use in the general switched
       telephone network". ITU-T Recommendation V.21, November 1988.

  VIII.International Telecommunication Union (ITU), "600/1200-baud modem
       standardized for use in the general switched telephone network".
       ITU-T Recommendation V.23, November 1988.

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Internet-Draft         Framework for ToIP using SIP          April 2008

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