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Versions: (draft-vanwijk-sipping-toip) 00 01 02 03 04 05 06 07 08 09 RFC 5194

       Internet Engineering Task Force                 SIPPING WG
       Internet Draft
       Document: <draft-ietf-sipping-toip-01.txt>      A. van Wijk (editor)
       July 18 2005                                    Viataal
       Expires: January 17 2006
        Framework of requirements for real-time text conversation using SIP.
       Status of this Memo
          By submitting this Internet-Draft, each author represents that any
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       Copyright Notice
          Copyright (C) The Internet Society (2005).
          This document provides the framework of requirements for real-time
          character-by-character interactive text conversation over the IP
          network using the Session Initiation Protocol and the Transport
          Protocol for Real-Time Applications. It discusses requirements for
          real-time Text-over-IP telephony as well as interworking between
          Text-over-IP telephony and existing text telephony on the PSTN and
          other networks.
       A. van Wijk                                           [Page 1 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
       Table of Contents
          1. Introduction                                              3
          2. Scope                                                     3
          3. Terminology                                               3
          4. Definitions                                               4
          5. Framework Description                                     5
          5.1. Background                                              5
          5.2. Requirements for ToIP                                   6
          5.3. Use of SIP and RTP                                      6
          5.4. Requirements for ToIP Interworking                      9
          6. Detailed requirements for Text-over-IP                    9
          6.1. Pre-Call Requirements                                   10
          6.2 Basic Point-to-Point Call Requirements                   10
          6.2.1 Session Setup                                          10
          6.2.2 Addressing                                             11
          6.2.3 Alerting and session progress presentation             11
          6.2.4 Call Negotiations                                      12
          6.2.5 Answering                                              12
          6.2.6 Actions During Calls                                   13
          6.2.7 Additional session control                             14
          6.2.8 File storage                                           15
          6.3 Conference Call Requirements for ToIP User Agents        15
          6.4 Transport via RTP                                        15
          6.5 Character Set                                            16
          6.6 Transcoding                                              16
          6.7 Relay Services                                           16
          6.8 Emergency services                                       17
          6.9 User Mobility                                            17
          6.10 Confidentiality and Security                            17
          7. Interworking Requirements for ToIP                        17
          7.1 ToIP Interworking Gateway Services                       17
          7.2 ToIP and PSTN/ISDN Text-Telephony                        18
          7.3 ToIP and Cellular Wireless circuit switched Text-Telephony
          7.3.1 "No-gain"                                              19
          7.3.2 Cellular Text Telephone Modem (CTM)                    19
          7.3.3 "Baudot mode"                                          19
          7.3.4 Data channel mode                                      19
          7.3.5 Common Text Gateway Functions                          19
          7.4 ToIP and Cellular Wireless ToIP                          20
          7.5 Instant Messaging Support                                20
          7.6 IP Telephony with Traditional RJ-11 Interfaces           21
          7.7 Multi-functional gateways                                22
          7.8 ToIP interoperability with PSTN text telephones.         22
          7.9 Gateway Discovery                                        22
          8. Afterword                                                 23
          9. Security Considerations                                   23
          10. Authors Addresses                                        24
          11. References                                               25
          11.1 Normative                                               25
          11.2 Informative                                             27
       A. van Wijk                                           [Page 2 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
       1. Introduction
          For many years, text has been in use as a medium for
          conversational, interactive dialogue between users in a similar
          way as voice telephony is used. Such interactive text is different
          from messaging and semi-interactive solutions like Instant
          Messaging in that it offers an equivalent conversational
          experience to users that cannot, or do not wish to, use voice. It
          therefore meets a different set of requirements than other text-
          based solutions already available on IP networks.
          Traditionally, deaf, hard of hearing and speech-impaired people
          are amongst the most proliferate users of conversational,
          interactive text, but because of its interactivity, it is becoming
          popular amongst mainstream user groups as well.
          This document describes how existing IETF protocols can be used to
          implement a Text-over-IP solution (ToIP). This ToIP framework is
          specifically designed to be compatible with Voice-over-IP
          environments, as well as meeting the userÆs requirements,
          including those of deaf, hard of hearing and speech-impaired users
          as described in RFC3351 [21].
          The Session Initiation Protocol (SIP) is the protocol of choice
          for control of Multimedia IP telephony and Voice-over-IP (VoIP)
          communications. It offers all the necessary control and signaling
          required for the ToIP framework.
          The Real-Time Transport Protocol (RTP) is the protocol of choice
          for real-time data transmission, and its use for interactive text
          payloads is described in RFC4103 [5].
          This document defines a framework for ToIP to be used either by
          itself or as part of integrated services, including Total
       2. Scope
          The primary scope of this document is to define a framework for
          the implementation of ToIP, either stand-alone or as a part of
          wider services, including Total Conversation. In general, the
          scope is:
          a. Description of ToIP using SIP and RTP;
          b. Requirements of Real-time, interactive text;
          c. Requirements for ToIP interworking.
          The subsequent sections describe those requirements in detail.
       3. Terminology
          In this document, the key words "MUST", "MUST NOT", "REQUIRED",
          RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as
          described in BCP 14, RFC 2119 [2] and indicate requirement levels
          for compliant implementations.
       A. van Wijk                                           [Page 3 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
       4. Definitions
          Audio bridging - a function of a gateway or relay service that
          enables an audio path through the service between the users
          involved in the call.
          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 can be introduced into the presented media.
          Interactive 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.
          TTY û alternative designation for a text telephone, often used in
          USA, see textphone. Also called TDD, Telecommunication Device for
          the Deaf.
          Textphone û also ôtext telephoneö. A terminal device that allows
          end-to-end real-time, interactive text communication. A variety of
          textphone protocols exists world-wide, both in the PSTN and other
          networks. 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 a gateway service that enables the
          flow of text through the service between the users involved in the
          Text gateway - a multi functional gateway that is able to
          transcode between different forms of text transport methods, e.g.,
          between ToIP in IP networks and Baudot text telephony in the PSTN.
          Text telephony û analog textphone services
          Text Relay Service - a third-party or intermediary that enables
          communications between deaf, hard of hearing and speech-impaired
          people, and voice telephone users by translating between voice and
          text in a call.
          Transcoding Services - services of a third-party user agent that
          transcodes one stream into another. Transcoding can be done by
          human operators, in automated manner or a combination of both
          methods. Text Relay Services are examples of a transcoding service
          between text and audio.
          Total Conversation - A multimedia service offering real time
          conversation in video, text and voice according to interoperable
          standards. All media flow in real time. Further defined in ITU-T
          F.703 Multimedia conversational services description.
       A. van Wijk                                           [Page 4 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
          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)
          CDMA   Code Division Multiple Access
          CTM    Cellular Text Telephone Modem
          GSM    Global System of Mobile Communication
          ISDN   Integrated Services Digital Network
          ITU-T  International Telecommunications Union-Telecommunications
          standardisation Sector
          PSTN   Public Switched Telephone Network
          SIP    Session Initiation Protocol
          TDD    Telecommunication Device for the Deaf
          TDMA   Time Division Multiple Access
          ToIP   Text over Internet Protocol
          UTF-8  Universal Transfer Format-8
       5. Framework Description
       5.1. Background
          The main purpose of this document is to provide a framework
          description for the implementation of real-time, interactive text
          based conversational services over IP networks, known as Text-
          over-IP (ToIP).
          This framework uses existing standards that are already commonly
          used for voice based conversational services on IP networks. In
          particular, the ToIP framework uses the Session Initiation
          Protocol (SIP) [3] to set up, control and tear down the
          connections between users.
          Media is transported using the Real-Time Transport Protocol (RTP)
          in the manner described in RFC4103.
          This framework allows for implementation of services that meet the
          requirement of providing a text-based conversational service,
          equivalent to voice based telephony. In particular, ToIP offers an
          IP equivalent of text telephony services as used by deaf, hard of
          hearing and speech-impaired individuals.
          In addition, real-time text conversations can be combined with
          other conversational services using different media like video or
          By using SIP, ToIP allows participants to negotiate all media
          including real-time text conversation[4, 5]. This is a highly
          desirable function for all IP telephony users, but essential for
          deaf, hard of hearing, or speech impaired people who have limited
          or no use of the audio path of the call.
          It is important to understand that real-time text conversations
          are significantly different from other text-based communications
       A. van Wijk                                           [Page 5 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
          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 takes place, thus providing the same mode of
          interaction as voice telephony does for hearing people. Store-and-
          forward systems like email or messaging on mobile networks or non-
          streaming systems like instant messaging are unable to provide
          that functionality.
       5.2. Requirements for ToIP
          In order to make ToIP the equivalent of what voice is to hearing
          people, it needs to offer equivalent features in terms of
          conversationality as voice telephony provides to hearing people.
          To achieve that, ToIP MUST:
          a. Offer real-time presentation of the conversation;
          b. Provide simultaneous transmission in both directions;
          c. Provide interoperability with text conversation features in
          other networks, for instance the PSTN, accepting functional
          limitations that will occur during interoperation.
          d. Not prevent other media, like audio and video, to be used in
          conjunction with ToIP.
          Users might want to use multiple modes of communication during the
          conversation, either at the same time or by switching between
          modes, e.g., between text and audio for example. Native ToIP
          services MUST ensure that the text interface is always available.
          When communicating via a gateway to other networks and protocols,
          the service SHOULD support all the functionality for alternating
          or simultaneous use of modalities as offered by the destination
          ToIP will often be used to access a relay service [I], allowing
          text users to communicate with voice users. With relay services,
          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 as small as possible. In
          particular, buffering of whole lines of text MUST NOT be used.
       5.3. Use of SIP and RTP
          ToIP services MUST use the Session Initiation Protocol (SIP) [3]
          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.
          Thus, participants are allowed to negotiate on a set of compatible
          media types with session descriptions used in SIP invitations. A
          ToIP service MUST always support at least one Text media type.
       A. van Wijk                                           [Page 6 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
          ToIP services MUST use the Real-Time Transport Protocol (RTP)
          according to the specification of RFC4103 for the transport of
          text between participants, which implements T.140 on IP networks.
          The standardized T.140 real-time text conversation [4], in
          addition to audio and video communications, will be a valuable
          service to many, including on non-IP networks. Real-time text can
          be expressed as a part of the session description in SIP and is a
          useful subset of Total Conversation.
          The ToIP specification describes a framework for using the T.140
          text conversation in SIP as a part of the multimedia session
          establishment in real-time over a SIP network.
          If the User Agents of different participants indicate that there
          is an incompatibility between their capabilities to support
          certain media types, e.g. one terminal only offering T.140 over IP
          as described in RFC4103 and the other one only supporting audio,
          the user might want to invoke a transcoding services.
          Examples of possible scenarios for including a relay service in
          the conversation are: speech-to-text (STT), text-to-speech (TTS),
          text bridging after conversion from speech, audio bridging after
          conversion from text, etc.
          The session description protocol (SDP) [6] used in SIP to describe
          the session is used to express these attributes of the session
          (e.g., uniqueness in media mapping for conversion from one media
          to another for each communicating party).
          Real-time text can also be presented in conjunction with other
          media like video and audio, as for example in Total Conversation
          User Agents providing ToIP functionality SHOULD provide suitable
          alerting, specifically offering visual and/or tactile alerting so
          that deaf and hard of hearing users can use them.
          The SIP abilities to set up text conversation sessions from any
          location, as well as privacy and security provisions SHOULD be
          implemented in ToIP services.
          Where ToIP is used in conjunction with other media, exposure of
          SIP functions through the User Interface MUST be available in
          equivalent fashion for all supported media. In other words, where
          certain SIP call control functions are available for the audio
          media part of the session, these functions MUST also be supported
          for the text media part of the same session.
          Any ToIP implementation MUST also allow invocation and use of
          relevant transcoding services where these are available. This can
          be achieved through application of SIP techniques for different
       A. van Wijk                                           [Page 7 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
          session establishment models [7]: Third party call control [8] and
          Conference Bridge model [9].
          Both point-to-point and multipoint communication need to be
          defined for the session establishment using T.140 text
          conversation. In addition, ToIP services SHOULD support
          interworking with text telephony [10].
          The general framework for ToIP can be described as follows:
          a. Session setup, modification and teardown procedures for point-
             to-point and multimedia calls
          b. Registration procedures and address resolutions
          c. Registration of user preferences
          d. Negotiation procedures for device capabilities
          e. Discovery and invocation of transcoding/translation services
          between the media in the call
          f. Different session establishment models for transcoding /
          translation services invocation: Third party call control and
          conference bridge model
          g. Uniqueness in media mapping to be used in the session for
          conversion from one media to another by the transcoding /
          translation server for each communicating party
          h. Media bridging services for T.140 real-time text as described
          in RFC4103, audio, and video for multipoint communications
          i. Transparent session setup, modification, and teardown between
          text conversation capable and voice/video capable devices
          j. Support of text media transport using T.140 over RTP as laid
          out in RFC 4103 [4]
          k. Signaling of status information, call progress and the like in
          a suitable manner, bearing in mind the user may have a hearing
          l. T.140 real-time text presentation mixing with voice and video
          m. T.140 real-time text conversation sessions using SIP, allowing
          users to move from one place to another
          n. User privacy and security for sessions setup, modification, and
          teardown as well as for media transfer
          o. Interoperability between T.140 conversations and analogue text
       A. van Wijk                                           [Page 8 of 28]
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          p. Routing of emergency calls according to national or regional
          policy to the same level of a voice call.
       5.4. Requirements for ToIP 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, which also defines modem detection sequences for the
          different text terminals. The full modem capability exchange
          between two wildly different terminals can take more than one
          minute to complete if both terminals have a common text
          To resolve international analog textphone incompatibilities, text
          telephone gateways MUST transcode incoming analog signals into
          T.140 and vice versa. The modem capability exchange time is then
          also reduced, since V.18 allows the sequence of protocol discovery
          to be customized. Hence, the text telephone gateways will assume
          the analog text telephone protocol used in the region the gateway
          is located. For example, in the USA, Baudot might be tried as the
          initial protocol. If negotiation for Baudot fails, the full modem
          capability exchange will then take place. In contrast, in the UK,
          ITU-T V.21 might be the first choice.
       6. Detailed requirements for Text-over-IP
          ToIP services MUST use SIP for call control and signaling.
          A ToIP user may wish to call another ToIP user, or join a
          conference call involving several users. He or she may, also, wish
          to initiate or join a multimedia call, such as a Total
          Conversation call.
          There may be some need for pre-call setup e.g. storing
          registration information in the SIP registrar to provide
          information about how a user can be contacted. This will allow
          calls to be set up rapidly and with proper routing and addressing.
          Similarly, there are requirements that need to be satisfied during
          call set up when other media are preferred by a user. For
          instance, some users may prefer to use audio while others want to
          use text as their preferred modality. In this case, transcoding
          services might be needed for text-to-speech (TTS) and speech-to-
          text (STT). The requirements for transcoding services need to be
          negotiated in real-time to set up the session.
          The subsequent subsections describe some of these requirements in
       A. van Wijk                                           [Page 9 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
       6.1. Pre-Call Requirements
          The need to use ToIP as a medium of communications can be
          expressed by users during registration time. Two situations need
          to be considered in the pre-call setup environment:
          a. User Preferences: It MUST be possible for a user to indicate a
          preference for ToIP by registering that preference with a SIP
          server that is part of the ToIP service.
          b. Server to support User Preferences: SIP servers that are part
          of ToIP services MUST have the capability to act on users
          preferences for ToIP to accept or reject the call, based on the
          user preferences defined during the pre-call setup registration
          time. For example, if the user is called by another party, and it
          is determined that a transcoding server is needed, the call MUST
          be re-directed or otherwise handled accordingly.
       6.2 Basic Point-to-Point Call Requirements
          The point-to-point call will take place between two parties. The
          requirements are described in subsequent sub-sections. They assume
          that one or both of the communicating parties will indicate ToIP
          as a possible or preferred medium for conversation using SIP in
          the session setup.
       6.2.1 Session Setup
          Users will set up a session by identifying the remote party or the
          service they will want to connect to. However, conversations could
          be started using a mode other than ToIP. For instance, the
          conversation might be established using audio and the user could
          subsequently elect to switch to text, or add text as an additional
          modality, during the conversation. Systems supporting ToIP MUST
          allow users to select any of the supported conversation modes at
          any time, including mid-conversation.
          Systems SHOULD allow the user to specify a preferred mode of
          communication, with the ability to fall back to alternatives that
          the user has indicated are acceptable.
          If the user requests simultaneous use of text and audio, and this
          is not possible either because the system only supports alternate
          modalities or because of resource management on the network, the
          system MUST try to establish a text-only communication. The user
          MUST be informed of this change throughout the process, either in
          text or in a combination of modalities that MUST include text.
          Session setup, especially through gateways to other networks, MAY
          require the use of specially formatted addresses or other
          mechanisms for invoking gateways.
       A. van Wijk                                           [Page 10 of 28]
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          The following features MAY need to be implemented to facilitate
          the session establishment using ToIP:
          a. Caller Preferences: SIP headers (e.g., Contact) can be used to
          show that ToIP is the medium of choice for communications.
          b. Called Party Preferences: The called party being passive can
          formulate a clear rule indicating how a call should be handled
          either using ToIP as a preferred medium or not, and whether a
          designated SIP proxy needs to handle this call or it is handled in
          the SIP user agent (UA).
          c. SIP Server support for User Preferences: SIP servers can also
          handle the incoming calls in accordance to preferences expressed
          for ToIP. The SIP Server can also enforce ToIP policy rules for
          communications (e.g. use of the transcoding server for ToIP).
       6.2.2 Addressing
          The SIP [3] addressing schemes MUST be used for all entities. For
          example SIP URL and Tel URL will be used for caller, called party,
          user devices, and servers (e.g., SIP server, Transcoding server).
          The right to include a transcoding service MUST NOT require user
          registration in any specific SIP registrar, but MAY require
          authorisation of the SIP registrar in the service.
       6.2.3 Alerting and session progress presentation
          User Agents supporting ToIP MUST have an alerting method (e.g.,
          for incoming calls) that can be used by deaf and hard of hearing
          people or provide a range of alternative, but equivalent, alerting
          methods that are suitable for all users, regardless of their
          abilities and preferences.
          It should be noted that general alerting systems exist, and one
          common interface for triggering the alerting action is a contact
          closure between two conductors.
          Among the alerting options are alerting by the User AgentÆs User
          Interface and specific alerting user agents registered to the same
          registrar as the main user agent.
          If present, identification of the originating party (for example
          in the form of a URL or CLI) MUST be clearly presented to the user
          in a form suitable for the user BEFORE answering the request. When
          the invitation to initiate a conversation involving ToIP
          originates from a gateway, this MAY be signaled to the user.
          During a conversation that includes ToIP, status and session
          progress information MUST be provided in text. That information
          MUST be equivalent to session progress information delivered in
          any other format, for example audio. Users MUST be able to manage
       A. van Wijk                                           [Page 11 of 28]
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          the session and perform all session control functions based on the
          textual session progress information.
          The user MUST be informed of any change in modalities.
          Session progress information SHOULD use simple language as much as
          possible so that as many users as possible can understand it. The
          use of jargon or ambiguous terminology SHOULD be avoided at all
          times. It is RECOMMENDED to let text information be used together
          with icons symbolising the items to be reported.
          There MUST be a clear indication, both visually as well as audibly
          whenever a session gets connected or disconnected. The user SHOULD
          never be in doubt as to what the status of the connection is, even
          if he/she is not able to use audio feedback or vision.
          In summary, it SHOULD be possible to observe visual or tactile
          indicators about:
          - Call progress
          - Availability of text, voice and video channels
          - Incoming call
          - Incoming text
          - Typed and transmitted text
          - Any loss in incoming text.
       6.2.4 Call Negotiations
          The Session Description Protocol (SDP) used in SIP [3] provides
          the capabilities to indicate ToIP as a media in the call setup.
          RFC 4103 [5] provides the RTP payload type text/t140 for support
          of ToIP which can be indicated in the SDP as a part of SDP INVITE,
          OK and SIP/200/ACK for media negotiations. In addition, SIPÆs
          offer/answer model can also be used in conjunction with other
          capabilities including the use of a transcoding server for
          enhanced call negotiations [7,8,9].
       6.2.5 Answering
          Systems SHOULD provide a best-effort approach to answering
          invitations for session set-up and users should be kept informed
          at all times about the progress of session establishment. On all
          systems that both inform users of session status and support ToIP,
          this information MUST be available in text, and MAY be provided in
          other visual media.
      Answering Machine
          Systems for ToIP MAY support an auto-answer function, equivalent
          to answering machines on telephony networks. 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
       A. van Wijk                                           [Page 12 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
          MAY support much larger storage. It is RECOMMENDED that systems
          support storage of at least 20 incoming messages of up to 16000
          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 MUST be
          allowed to intervene during any stage of the answering machine and
          take control of the session.
       6.2.6 Actions During Calls
          Certain actions need to be performed for the ToIP conversation
          during the call and these actions are described briefly as
          a. Text transmission SHALL be done character by character as
          entered, or in small groups transmitted so that no character is
          delayed between entry and transmission by more than 300
          b. The text transmission SHALL allow a rate of at least 30
          characters per second so that human typing speed as well as speech
          to text methods of generating conversation text can be supported.
          c. After text connection is established, the mean end-to-end delay
          of characters SHALL be less than two seconds, measured between two
          ToIP users. This requirement is valid as long as the text input
          rate is lower or equal to the text reception and display rate.
          d. The character corruption rate SHALL be less than 1% in
          conditions where users experience the quality of voice
          transmission to be low but useable. This is in accordance with
          ITU-T F.700 Annex A.3 quality level T1.
          e. When interoperability functions are invoked, there may be a
          need for intermediate storage of characters before transmission to
          a device receiving slower than the typing speed of the sender.
          Such temporary storage SHALL be dimensioned to adjust for
          receiving at 30 characters per second and transmitting at 6
          characters per second during at least 4 minutes [less than 3k
          f. To enable the use of international character sets the
          transmission format for text conversation SHALL be UTF-8, in
          accordance with ITU-T T.140.
          g. If text is detected to be missing after transmission, there
          SHALL be an indication in the text marking the loss. For 7 bit
          terminals this loss MAY be marked as an apostrophe: Æ.
          g. When used from a terminal designed for PSTN text telephony, or
          in interworking with such a terminal, ToIP shall enable
       A. van Wijk                                           [Page 13 of 28]
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          alternating between text and voice in a similar manner as the PSTN
          text telephone handles this mode of operation. (This mode is often
          called VCO/HCO in the USA and the UK).
          i. When display of the conversation on end user equipment is
          included in the design, display of the dialogue SHALL be made so
          that it is easy to read text belonging to each party in the
      Text and other Media Handling Between ToIP User Agents
          The following requirements are valid for media handling during
          a. When used between User Agents designed for ToIP, it SHALL be
          possible to send and receive text simultaneously.
          b. When used between User Agents that support ToIP, it SHALL be
          possible to send and receive text simultaneously with the other
          media (text, audio and/or video) supported by the same terminals.
          c. It SHOULD be possible to know during the call that ToIP is
          available, even if it is not invoked at call setup (only voice
          and/or video is used for example). To disable this, the user must
          disable the use of ToIP. This is possible during registration at
          the REGISTRAR.
      Call Action with Native ToIP User Agents
          a. It SHOULD be possible to answer a call with text capabilities
          b. It MAY be possible to use video simultaneously with the other
          media in the call.
          c. It MUST be possible to answer a call in voice or video without
          text enabled, and add text later in the call.
          d. It MUST be possible to disconnect the call.
          e. It SHOULD be possible to invoke multi-party calls.
          f. It MUST be possible to transfer the call.
       6.2.7 Additional session control
          Systems that support additional session control features, for
          example call waiting, forwarding, hold etc on voice calls, MUST
          offer equivalent functionality for text calls.
       A. van Wijk                                           [Page 14 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
       6.2.8 File storage
          Systems that support ToIP MAY save the text conversation to a
          file. This SHOULD be done using a standard file format. For
          example: UTF8 text file in XML format including record timestamp,
          party and the text conversation.
       6.3 Conference Call Requirements for ToIP User Agents
          The conference call requirements deal with multipoint conferencing
          calls where there will be at least one or more ToIP capable
          devices along with other end user devices where the total number
          end user devices will be at least three.
          It SHOULD be possible to use the text medium in conference calls,
          in a similar way as the audio is handled and the video is
          displayed. Text in conferences can be used both for letting
          individual participants use the text medium (for example, for
          sidebar discussions in text while listening to the main conference
          audio), as well as for central support of the conference with real
          time text interpretation of speech.
       6.4 Transport via RTP
          ToIP uses RTP as the default transport protocol for transmission
          of real-time text via medium text/t140 as specified in RFC 4103
          The redundancy method of RFC 4103 [5] SHOULD be used for making
          text transmission reliable.
          Text capability MUST be announced in SDP by a declaration in line
          with this example:
               m=text 11000 RTP/AVP 98 100
               a=rtpmap:98 t140/1000
               a=rtpmap:100 red/1000
               a=fmtp:100 98/98/98
          Characters SHOULD be buffered for transmission and transmitted
          every 300 ms.
          By having this single coding and transmission scheme for real time
          text defined, in the SIP call 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, and RFC 4103 [5] transport MUST be used
          as both the default as well as the fallback transport.
       A. van Wijk                                           [Page 15 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
       6.5 Character Set
          a. ToIP services MUST use UTF-8 encoding as specified in ITU-T
          T.140 [12].
          b. ToIP SHOULD handle characters with editing effect such as new
          line, erasure and alerting during session as specified in ITU-T
       6.6 Transcoding
          Transcoding of text may need to take place in gateways between
          ToIP and other forms of text conversation. For example to connect
          to a PSTN text telephone.
       6.7 Relay Services
          The relay service acts as an intermediary between two or more
          callers using different media or different media encoding schemes.
          The basic text relay service allows a translation of speech to
          text and text to speech, which enables hearing and speech impaired
          callers to communicate with hearing callers. Even though this
          document focuses on ToIP, we want to remind readers that there
          exist other relay services like, for example, speech to sign
          language and vice versa using video.
          It is RECOMMENDED that ToIP implementations make the invocation
          and use of relay services as easy as possible. It MAY happen
          automatically when the call is being set up based on any valid
          indication or negotiation of supported or preferred media types. A
          transcoding framework document using SIP [7] 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 framework.
          Adding or removing a relay service MUST be possible without
          disrupting the current call.
          When setting up a call, the relay service MUST be able to
          determine the type of service requested (e.g., speech to text or
          text to speech), to indicate if the caller wants voice carry over,
          the language of the text, the sign language being used (in the
          video stream), etc.
          It SHOULD be possible to route the call to a preferred relay
          service even if the user makes the call from another region or
          network than usually used.
       A. van Wijk                                           [Page 16 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
       6.8 Emergency services
          Access to emergency services using ToIP SHOULD provide an
          equivalent service to the one offered by other supported media,
          like audio.
       6.9 User Mobility
          ToIP User Agents SHOULD use the same mechanisms as other SIP User
          Agents to resolve mobility issues. It is RECOMMENDED to use a SIP-
          address for the users, resolved by a SIP REGISTRAR, to enable
          basic user mobility. Further mechanisms are defined for the 3G IP
          multimedia systems.
       6.10 Confidentiality and Security
          User confidentiality and privacy need to be met as described in
          SIP [3]. For example, nothing should reveal the fact that the user
          of ToIP is a person with a disability unless the user prefers to
          make this information public. If a transcoding server is being
          used, this SHOULD be transparent. Encryption SHOULD be used on
          end-to-end or hop-by-hop basis as described in SIP [3] and SRTP
          Authentication needs to be provided for users in addition to the
          message integrity and access control.
          Protection against Denial-of-service (DoS) attacks needs to be
          provided considering the case that the ToIP users might need
          transcoding servers.
       7. Interworking Requirements for ToIP
          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 systems. This section describes requirements on this
          interoperability, especially for the PSTN text telephony to ensure
          full backward interoperability with ToIP.
       7.1 ToIP Interworking Gateway Services
          Interactive texting facilities exist already in various forms and
          on various networks. On the PSTN, it is commonly referred to as
          text telephony.
          Simultaneous or alternating use of voice and text is used by a
          large number of users who can send voice, but must receive text or
          who can hear but must send text due to a speech disability.
       A. van Wijk                                           [Page 17 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
       7.2 ToIP and PSTN/ISDN Text-Telephony
          On PSTN networks, transmission of interactive text takes place
          using a variety of codings and modulations, including ITU-T V.21
          [II], Baudot, DTMF, V.23 [III] and others. Many difficulties have
          arisen as a result of this variety in text telephony protocols and
          the ITU-T V.18 [10] standard was developed to address some of
          these issues.
          ITU-T-V.18 [10] 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.
          In order to allow systems and services based on ToIP to
          communicate with PSTN text telephones, text gateways are the
          recommended approach. These gateways MUST use the ITU-T V.18 [10]
          standard at the PSTN side.
          Buffering MUST be used to support different transmission rates. At
          least 1K buffer MUST be provided. A buffer of at least 2K
          characters is RECOMMENDED. In addition, the gateway MUST provide a
          minimum throughput of at least 30 characters/second or the highest
          speed supported by the PSTN text telephony protocol side,
          whichever is the lowest.
          PSTN-ToIP gateways MUST allow alternating use of text and voice.
          PSTN and ISDN to ToIP gateways that receive CLI information from
          the originating party MUST pass this information to the receiving
          party as soon as possible.
          Priority MUST be given to calls labeled as emergency calls.
       7.3 ToIP and Cellular Wireless circuit switched Text-Telephony
          Cellular wireless (or Mobile) 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.
          Alternative means of transferring the Text telephony data have
          been developed when TTY services over cellular was mandated by the
          FCC in the USA. They are a) "No-gain" codec solution, b) the
          Cellular Text Telephony Modem (CTM) solution and c) "Baudot mode"
          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.
       A. van Wijk                                           [Page 18 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
       7.3.1 "No-gain"
          The "No-gain" text telephone transporting technology uses
          specially modified EFR [15] and EVR [16] speech vocoders in both
          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 the CDMA and TDMA mobile
          technologies and the US Baudot type of text telephones.
       7.3.2 Cellular Text Telephone Modem (CTM)
          CTM [17] is a technology independent modem technology that
          provides the transport of text telephone characters at up to 10
          characters/sec using modem signals that are at or below 1 kHz and
          uses a highly redundant encoding technique to overcome the fading
          and cell changing losses. On any interface that uses analog
          transmission, half-duplex operation must be supported as the
          "send" and "receive" modem frequencies are identical. The use of
          CTM may have to be modified slightly to support half-duplex
       7.3.3 "Baudot mode"
          This term is often used by cellular terminal suppliers for a GSM
          cellular phone mode that allows TTYs to operate into a cellular
          phone and to communicate with a fixed line TTY.
       7.3.4 Data channel mode
          Many mobile terminals allow the use of the data channel to
          transfer data in real-time. Data rates of 9600 bit/s are usually
          supported on the mobile network. Gateways or the interworking
          function provides interoperability with PSTN textphones.
       7.3.5 Common Text Gateway Functions
          Text gateways MUST cover the differences that result from
          different text protocols. The protocols to be supported will
          depend on the service requirements of the Gateway.
          Different data rates of different protocols MAY require text
          Interoperation of half-duplex and full-duplex protocols MAY
          require text buffering and some intelligence to determine when to
          change direction when operating in half-duplex.
          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
       A. van Wijk                                           [Page 19 of 28]
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          A text gateway MUST be able to route text calls to emergency
          service providers when any of the recognised emergency numbers
          that support text communications for the country or region are
          called eg. "911" in USA and "112" in Europe. Routing text calls to
          emergency services MAY require the use of a transcoding service.
          A text gateway MUST act as a SIP User Agent on the IP side.
       7.4 ToIP and Cellular Wireless ToIP
          ToIP MAY be supported over the cellular wireless packet switched
          service. It interfaces to the Internet. For 3GPP 3G services, the
          support is described to use ToIP in 3G TS 26.235 [20].
          A text gateway with cellular wireless packet switched services
          MUST be able to route text calls into emergency service providers
          when any of the recognized emergency numbers that support text
          communication for the country are called.
       7.5 Instant Messaging Support
          Many people use Instant Messaging to communicate via the Internet
          using text. Instant Messaging transfers blocks of text rather than
          streaming as is used by ToIP. As such, it is not a replacement for
          ToIP and in particular does not meet the needs for real time
          conversations of deaf, hard of hearing and speech-impaired users
          as defined in RFC 3351 [21]. It is unsuitable for communications
          through a relay service [I]. The streaming character of ToIP
          provides a better user experience and, when given the choice,
          users often prefer ToIP.
          However, since some users might only have Instant Messaging
          available, text gateways MAY be developed 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, such gateways need to
          transform between these two formats. Text gateways for
          interworking between Instant Messaging and ToIP MUST concatenate
          individual characters originating at the ToIP side into blocks of
          text and:
          a. 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.
          b. When a new line 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.
       A. van Wijk                                           [Page 20 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
          c. 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.
          It is RECOMMENDED that during the session, both users are
          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 signaling
          to the Instant Messaging side when characters start being
          received, or at the beginning of the conversation.
          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 per 5 seconds that pass a . (dot) can be
          Those solutions will reduce the difficulties between a streaming
          versus blocked text.
          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 extremely similar. After all, the character
          input, the character display, Internet connectivity and SIP stack
          are the same for Instant Messaging (SIMPLE) and ToIP.
          Devices that implement Instant Messaging SHOULD implement ToIP as
          described in this document.
       7.6 IP Telephony with Traditional RJ-11 Interfaces
          Analogue adapters using SIP based IP communication and RJ-11
          connectors for connecting traditional PSTN devices (ATA box)
          SHOULD enable connection of legacy PSTN text telephones [18].
          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 [5], in a
          similar way as it provides interoperability for voice calls. If a
          call is set up and text/t140 capability is not declared by the
          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.
          NOTE: It is preferred that such analogue adaptors do use RFC 4103
          [5] 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 dropping characters in the textphone
          conversation or even not allowing the textphones to connect with
          each other.
       A. van Wijk                                           [Page 21 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
       7.7 Multi-functional gateways
          In practice many interworking gateways will be implemented as
          gateways that combine different functions. As such, a text gateway
          could be build 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 MUST 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 real-time ToIP MUST support the following
          interworking functions:
          - PSTN text telephony to real-time ToIP.
          - PSTN text telephony to Instant Messaging.
          - Instant Messaging to real-time ToIP.
       7.8 ToIP interoperability with PSTN text telephones.
          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.
          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 10 646 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 missing text indicator in T.140, specified in T.140 amendment
          1, cannot be represented in the 7 bit character codes. Therefore
          these characters 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.
       7.9 Gateway Discovery
          ToIP requires a method to invoke a text gateway. As described
          previously in this draft, these text gateways MUST act as User
          Agents at the IP side. The capabilities of the text 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 only able to receive voice and streaming text, so the text
          gateway will only allow ToIP and audio.
          Examples of possible scenarios for discovery of the text gateway
       A. van Wijk                                           [Page 22 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
          - PSTN textphone users dial a prefix number before dialing out.
          - Separate text subscriptions, linked to the phone number or
          terminal identifier/ IP address.
          - Text capability indicators.
          - Text preference indicator.
          - Listen for V.18 modem modulation text activity in all calls.
          - Call transfer request by the called user.
          - Placing a call via the web, and using one of the methods
          described here
          - Text gateways with its own telephone number and/or SIP address.
          (This requires user interaction with the text gateway to place a
          - ENUM address analysis and number plan
          - Number or address analysis leads to the gateway for all PSTN
       8. Afterword
          The authors want to make it clear that ToIP is a way of allowing
          real-time, interactive text conversation between all users and is
          thus not only for the hearing and speech impaired users.
          The users may invoke the ToIP services for many different reasons.
          For example:
          - Noisy environment (e.g., in a machine room of a factory where
          listening is difficult)
          - Busy with another call and want to participate in two calls at
          the same time.
          - Text and/or speech recording services (e.g., text
          documentation/audio recording for legal/clarity/flexibility
          - Overcoming of language barriers through speech translation
          and/or transcoding services.
          - Hearing loss, tinnitus or deafness due to the aging process or
          any other reason.
          NOTE: In many of the above examples, text may accompany speech and
          could be displayed in a manner similar to subtitling in
          broadcasting environments or any other suitable manner.  This
          could occur for individuals who are hard of hearing and also for
          mixed calls with a hearing and deaf person listening to the call.
       9. Security Considerations
          There are no additional security requirements other than described
       A. van Wijk                                           [Page 23 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
       10. Authors Addresses
          The following people provided substantial technical and writing
          contributions to this document, listed alphabetically:
          Willem P. Dijkstra
          TNO Informatie- en Communicatietechnologie
          Postbus 15000
          9700 CD Groningen
          The Netherlands
          Tel: +31 50 585 77 24
          Fax: +31 50 585 77 57
          Email: willem.dijkstra@tno.nl
          Barry Dingle
          ACIF, 32 Walker Street
          North Sydney, NSW 2060 Australia
          Tel +61 (0)2 9959 9111
          Fax +61 (0)2 9954 6136
          TTY +61 (0)2 9923 1911
          Mob +61 (0)41 911 7578
          Email barry.dingle@bigfoot.com.au
          Guido Gybels
          Department of New Technologies
          RNID, 19-23 Featherstone Street
          London EC1Y 8SL, UK
          Tel +44(0)20 7294 3713
          Txt +44(0)20 7296 8019
          Fax +44(0)20 7296 8069
          Email: guido.gybels@rnid.org.uk
          Gunnar Hellstrom
          Omnitor AB
          Renathvagen 2
          SE 121 37 Johanneshov
          Phone: +46 708 204 288 / +46 8 556 002 03
          Fax:   +46 8 556 002 06
          Email: gunnar.hellstrom@omnitor.se
          Henry Sinnreich
          115 Broadhollow Rd
          Suite 225
          Melville, NY 11747
          Tel: +1.631.961.8950
       A. van Wijk                                           [Page 24 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
          Gregg C Vanderheiden
          University of Wisconsin-Madison
          Trace R & D Center
          1550 Engineering Dr (Rm 2107)
          Madison, Wi  53706
          Phone +1 608 262-6966
          FAX +1 608 262-8848
          Arnoud A. T. van Wijk
          Viataal (Dutch Institute for the Deaf)
          Research & Development
          Afdeling RDS
          Theerestraat 42
          5271 GD Sint-Michielsgestel
          The Netherlands.
          Email: a.vwijk@viataal.nl
       11. References
       11.1 Normative
          1. Bradner, S., "The Internet Standards Process -- Revision 3",
          BCP 9, RFC 2026, October 1996.
          2. Bradner, S., "Key words for use in RFCs to Indicate Requirement
          Levels", BCP 14, RFC 2119, March 1997
          3. 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.
          4. ITU-T Recommendation T.140, "Protocol for Multimedia
          Application Text Conversation (February 1998) and Addendum 1
          (February 2000).
          5. G. Hellstrom, "RTP Payload for Text Conversation, RFC 4103,
          June 2005.
          6. G. Camarillo, H. Schulzrinne, and E. Burger, "The Source and
          Sink Attributes for the Session Description Protocol," IETF,
          August 2003 - Work in Progress.
          7. G.Camarillo, "Framework for Transcoding with the Session
          Initiation Protocol" IETF June 2005 -  Work in progress.
          8. 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,
          June 2005.
       A. van Wijk                                           [Page 25 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
          9. G. Camarillo, "The SIP Conference Bridge Transcoding Model,"
          IETF, August 2003 - Work in Progress.
          10. ITU-T Recommendation V.18,"Operational and Interworking
          Requirements for DCEs operating in Text Telephone Mode," November
          11. "XHTML 1.0: The Extensible HyperText Markup Language: A
          Reformulation of HTML 4 in XML 1.0", W3C Recommendation. Available
          at http://www.w3.org/TR/xhtml1.
          12. Yergeau, F., "UTF-8, a transformation format of ISO 10646",
          RFC 2279, January 1998.
          13. 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.)
          14. Bell-103 300 bit/s modem.
          15. TIA/EIA/IS-823-A  "TTY/TDD Extension to TIA/EIA-136-410
          Enhanced Full Rate Speech Codec (must used in conjunction with
          16. TIA/EIA/IS-127-2 "Enhanced Variable Rate Codec, Speech Service
          Option 3 for Wideband Spread Spectrum Digital Systems. Addendum
          17. 3GPP TS26.226  "Cellular Text Telephone Modem Description"
          18. I. Butcher, S. Lass, D. Petrie, H. Sinnreich, and C.
          Stredicke, "SIP Telephony Device Requirements, Configuration and
          Data," IETF, February 2004 - Work in Progress.
          19.  Baugher, McGrew, Carrara, Naslund, Norrman, "The Secure Real
          Time Transport Protocol (SRTP)", RFC 3711, IETF, March 2004.
          20. IP Multimedia default codecs. 3GPP TS 26.235
          21. 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.
          22. J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with the
          Session Description Protocol (SDP)", RFC 3624, IETF, June 2002.
       A. van Wijk                                           [Page 26 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
       11.2 Informative
          I. A relay service allows the users to transcode between different
          modalities or languages. In the context of this document, relay
          services will often refer to text relays that transcode text into
          voice and vice-versa. See for example http://www.typetalk.org.
          II. 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.
          III. International Telecommunication Union (ITU), "600/1200-baud
          modem standardized for use in the general switched telephone
          network. ITU-T Recommendation V.23, November 1988.
          IV. Third Generation Partnership Project (3GPP), "Technical
          Specification Group Services and System Aspects; Cellular Text
          Telephone Modem; General Description (Release 5)". 3GPP TS 26.226
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       A. van Wijk                                           [Page 27 of 28]
       draft-ietf-sipping-ToIP-01.txt                        July 18 2005
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       A. van Wijk                                           [Page 28 of 28]

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