SIPPING Workgroup                               A. van Wijk (editor)
       Internet-Draft                                  Viataal
       Category: Informational
       Expires: March 6 2006                           September 7 2005
        Framework of requirements for real-time text conversation using SIP
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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 Real-Time
          Transport Protocol. It discusses requirements for real-time Text-
          over-IP as well as interworking between Text-over-IP and existing
          text telephony on the PSTN and other networks.
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       Table of Contents
       1. Introduction.....................................................3
       2. Scope............................................................4
       3. Terminology......................................................4
       4. Definitions......................................................4
       5. Framework Description............................................6
       5.1. General requirements for ToIP..................................6
       5.1.1 General ToIP Summary..........................................8
       5.2. General Requirements for ToIP Interworking.....................8
       5.2.1 PSTN Interworking.............................................9
       5.2.2 Cellular circuit switched Text-Telephony.....................10 Cellular "No-gain".........................................10 Cellular Text Telephone Modem (CTM)........................10 Cellular "Baudot mode".....................................11
       5.2.3 Cellular data channel mode...................................11
       5.2.4 Cellular Wireless ToIP.......................................11
       5.2.5 Instant Messaging Support....................................11
       6. Detailed requirements for ToIP..................................11
       6.1. Pre-Session Requirements......................................12
       6.2 Basic Point-to-Point Session Requirements......................12
       6.2.1 Session control..............................................12
       6.2.2 Text transport...............................................12
       6.2.3 Session Setup................................................13
       6.2.4 Addressing...................................................13
       6.2.5 Alerting.....................................................14
       6.2.6 Session information..........................................14
       6.2.7 Session progress information.................................14
       6.2.8 Session Negotiations.........................................15
       6.2.9 Answering....................................................15 Answering Machine..........................................15
       6.2.10 Actions During a Session....................................15 Text Transport............................................16 Handling Text and other Media.............................16
       6.2.11 Additional session control..................................17
       6.2.12 File storage................................................17
       6.3 Conference Session Requirements................................17
       6.4 Real-time Editing and User Alerting............................17
       6.5 Emergency services.............................................17
       6.6 User Mobility..................................................18
       6.7 Firewalls and NATs.............................................18
       7. Interworking Requirements for ToIP..............................18
       7.1 ToIP Interworking Gateway Services.............................18
       7.2 ToIP and PSTN/ISDN Text-Telephony Interworking.................18
       7.3 ToIP and Cellular Wireless ToIP................................19
       7.4 Instant Messaging Support......................................19
       7.5 Common Text Gateway Functions..................................20
       7.5.1 Protocol support.............................................20
       7.5.2 Relay buffer storage.........................................20
       7.5.3 Emergency calls through gateways.............................21
       7.5.4 Text Gateway Invocation......................................21
       7.6 Home Gateways or Analog Terminal Adapters......................21
       7.7 Multi-functional Combination gateways..........................22
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       7.8 Transcoding....................................................22
       7.9 Relay Services.................................................23
       7.9.1 Basic function of the relay service..........................23
       7.9.2 Invocation of relay services.................................23
       8. Security Considerations.........................................23
       9. Authors Addresses...............................................24
       10. References.....................................................25
       10.1 Normative references..........................................25
       10.2 Informative references........................................27
       1. Introduction
          For many years, 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 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 conversational, interactive
          text but, because of its interactivity, it is becoming popular
          amongst mainstream users 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 (VoIP)
          environments, as well as meeting the userÆs requirements,
          including those of deaf, hard of hearing and speech-impaired users
          as described in RFC3351 [19].
          The Session Initiation Protocol (SIP) is the protocol of choice
          for control of Multimedia communications and Voice-over-IP (VoIP)
          in particular. 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, multi-media services, including
          Total Conversation.
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       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. It defines the:
             a. Requirements of Real-time, interactive text;
             b. Requirements for ToIP interworking;
             c. Description of ToIP using SIP and RTP;
             d. Description of ToIP interworking with other text services.
       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.
       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.
          Cellular - Telephone systems based on radio transmission to become
          wireless. Also called Wireless or Mobile systems.
          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.
          Textphone û also "text telephone". A terminal device that allows
          end-to-end real-time, interactive 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 a gateway service that enables the
          flow of text through the service between the users involved in the
          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.
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          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.
          Text telephony û analog textphone service.
          Total Conversation - a multimedia service offering real time
          conversation in video, text and voice according to interoperable
          standards. All media flow in real time. (See ITU-T F.703
          "Multimedia conversational services".)
          Transcoding Services - services of a third-party user agent that
          transcodes one stream into another. Transcoding can be done by
          human operators, in an automated manner or a combination of both
          methods. Text Relay Services are examples of a transcoding service
          between text and audio.
          TTY û alternative designation for a text telephone or textphone,
          often used in USA. 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)
          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 of Mobile Communication
          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   Text over Internet Protocol
          UTF-8  Universal Transfer Format-8
          VCO/HCO Voice Carry Over/Hearing Carry Over
          VoIP   Voice over Internet Protocol
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       5. Framework Description
          This framework defines the requirements of a text-based
          conversational service that is the text equivalent of voice based
          telephony. Real-time text conversation can be combined with other
          conversational services like video or voice.
          ToIP also offers an IP equivalent of analog text telephony
          services as used by deaf, hard of hearing and speech-impaired
          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 takes place. This provides 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. In particular, they do not
          allow for smooth communication through a Text Relay Service.
          This framework uses existing standards that are already commonly
          used for voice based conversational services on IP networks. It
          uses the Session Initiation Protocol (SIP) to set up, control and
          tear down the connections between users whilst the media is
          transported using the Real-Time Transport Protocol (RTP) as
          described in RFC4103 [5].
          This framework is designed to meet the requirements of RFC3351
          [19]. As such, it offers a standardized way for offering text-
          based, conversational services that can be used as an equivalent
          to voice telephony by deaf, hard of hearing and speech-impaired
          SIP 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.
       5.1. General requirements for ToIP
          In order to make ToIP the text equivalent of voice services, it
          needs to offer equivalent features in terms of conversationality
          as voice telephony provides. To achieve that, ToIP needs to:
             a. Offer real-time presentation of the conversation;
             b. Provide simultaneous transmission in both directions;
             c. Support both point-to-point and multipoint communication;
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             d. Allow other media, like audio and video, to be used in
               conjunction with ToIP;
             e. Ensure that the text service is always available.
          Real-time text is a useful subset of Total Conversation defined in
          ITU-T F.703 [23]. Users could use multiple modes of communication
          during the conversation, either at the same time or by switching
          between modes, e.g., between text and audio.
          Users may invoke ToIP services for many different reasons:
          - 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/clarity/flexibility purposes).
          - 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, thus creating a need to
            replace or complement voice with text in conversational
          NOTE: In many of the above examples, text may accompany speech.
          The text could be displayed side by side, in a manner similar to
          subtitling in broadcasting environments, or in any other suitable
          manner.  This could occur for users who are hard of hearing and
          also for mixed media calls with both hearing and deaf people
          participating in the call.
          User Agents providing ToIP functionality need to provide suitable
          alerting indications, specifically offering visual and/or tactile
          alerting for deaf and hard of hearing users.
          The ability of SIP to set up conversation sessions from any
          location, as well as its privacy and security provisions, MUST be
          maintained by ToIP services.
          Where ToIP is used in conjunction with other media, exposure of
          SIP functions through the User Interface needs to be done in an
          equivalent manner 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. For example, call
          transfer must act on all media in the session.
          T.140 real-time text conversation [4], in addition to audio and
          video communications, is a valuable service for many users,
          including those on non-IP networks. T.140 also provides for real-
          time editing of the text.
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       5.1.1 General ToIP Summary
          The general requirements for ToIP are:
          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. Support of text media transport using T.140 over RTP as
             described in RFC 4103 [5]
          f. Signaling of status information, call progress and the like in
             a suitable manner, bearing in mind that the user may have a
             hearing impairment
          g. T.140 real-time text presentation mixing with voice and video
          h. T.140 real-time text conversation sessions using SIP, allowing
             users to move from one place to another
          i. User privacy and security for sessions setup, modification, and
             teardown as well as for media transfer
          j. Routing of emergency calls according to national or regional
             policy with the same level of functionality as a voice call.
       5.2. General Requirements for ToIP Interworking
          This section describes the general ToIP interworking requirements
          and gives some background information to many of the issues.
          There is a range of existing text services. There is also a range
          of network technologies that could support text services (see
          examples below). ToIP needs to provide interoperability with text
          conversation features in other networks, for instance the PSTN,
          and with some text messaging services.
          Text gateways are used for converting between different media
          types. They could be used between networks or within networks
          where different transport technologies are used.
          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 destination
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          Address information, both called and calling, SHOULD be
          transferred, and possibly converted, when interworking between
          different networks.
          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 minimal. In particular,
          buffering of whole lines of text will not meet character delay
          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 [5] and the other one only supporting
          audio, the user might want to invoke a transcoding service.
          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 general requirements for ToIP Interworking are:
          a. Interoperability between T.140 conversations [4] and analog
             text telephones
          b. Discovery and invocation of transcoding/translation services
             between the media in the call
          c. Different session establishment models for transcoding /
             translation services invocation: Third party call control and
             conference bridge model
          d. 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
          e. Media bridging services for T.140 real-time text, as described
             in RFC4103 [5], audio and video for multipoint communications
          f. Transparent session setup, modification, and teardown between
             text conversation capable devices and voice/video capable
          g. Buffering of text when interworking with media that transport
             text at different rates.
       5.2.1 PSTN Interworking
          Analog text telephony is cumbersome because of incompatible
          national implementations where interworking was never considered.
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          A large number of these implementations have been documented in
          ITU-T V.18 [10], 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
          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 [III] 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 [II] might be the first choice.
       5.2.2 Cellular 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
          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.
      Cellular "No-gain"
          The "No-gain" text telephone transporting technology uses
          specially modified EFR [13] and EVR [14] 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 [15] 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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      Cellular "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.
       5.2.3 Cellular 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 provide interoperability
          with PSTN textphones.
       5.2.4 Cellular Wireless ToIP
          ToIP could be supported over cellular wireless packet switched
          services that interface to the Internet. For 3GPP 3G services, the
          support is described to use ToIP in 3G TS 26.235 [18]. Low data
          rates and additional delays can affect performance.
       5.2.5 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 including those of deaf, hard of hearing and speech-
          impaired users as defined in RFC 3351 [19]. It is unsuitable for
          communications through a relay service [I]. The streaming nature
          of ToIP provides a more direct conversational user experience and,
          when given the choice, users may prefer ToIP.
          Text gateways could be developed to allow interworking between
          Instant Messaging systems and ToIP solutions.
       6. Detailed requirements for ToIP
          A ToIP user may wish to call another ToIP user, or join a
          conference session involving several users or initiate or join a
          multimedia session, such as a Total Conversation session.
          There may be some need for pre-session setup e.g. storing of
          registration 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
          Similarly, there are requirements that need to be satisfied during
          session set up when other media are preferred by a user. For
          instance, some users may indicate their preferred modality to be
          audio while others may indicate text. In this case, transcoding
          services might be needed for text-to-speech (TTS) and speech-to-
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          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
       6.1. Pre-Session Requirements
          The need to use 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 text by registering that preference with a SIP
             server that is part of the ToIP service.
          b. Server to support User Preferences: SIP servers that support
             ToIP services MUST have the capability to act on calling user
             preferences for text in order to accept or reject the session-,
             based on the called userÆs 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 MUST be re-directed
             or otherwise handled accordingly.
       6.2 Basic Point-to-Point Session Requirements
          A point-to-point session takes place between two parties. The
          requirements are described in subsequent sub-sections. They assume
          that one or both of the communicating parties will indicate text
          as a possible or preferred medium for conversation using SIP in
          the session setup.
       6.2.1 Session control
          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. The session description
          protocol (SDP) [6] used in SIP to describe the session is used to
          express the attributes of the session and to negotiate a set of
          compatible media types.
       6.2.2 Text transport
          A ToIP service MUST always support at least one Text media type.
          ToIP services MUST support the Real-Time Transport Protocol (RTP)
          [24] according to the specification of RFC4103 [5] for the
          transport of text between participants.
          RFC4103 describes the transmission of T.140 [4] on IP networks.
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       6.2.3 Session Setup
          Users will set up a session by identifying the remote party or the
          service they want to connect to. However, conversations could be
          started using a mode other than text. 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 constraints in the network, the system
          MUST try to establish communication with best effort. If the user
          has expressed a preference for text, establishment of a connection
          including text MUST have priority over other outcomes of the
          session setup.
          The following features MAY be implemented to facilitate the
          session establishment using ToIP:
          a. Caller Preferences: SIP headers (e.g., Contact)[24] 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 session should be
             handled either using 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: SIP servers can also
             handle the incoming sessions in accordance with preferences
             expressed for ToIP. The SIP Server can also enforce ToIP policy
             rules for communications (e.g. use of the transcoding server
             for ToIP).
       6.2.4 Addressing
          The SIP [3] addressing schemes MUST be used for all entities in a
          ToIP session. For example, SIP URLÆs or Tel URLÆs are 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 to invoke the service.
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       6.2.5 Alerting
          User Agents supporting ToIP MUST have an alerting method (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 abilities.
          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.
          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.
       6.2.6 Session information
          If present, identification of the originating party (for example
          in the form of a URL or a CLI) MUST be clearly presented to the
          user in a form suitable for the user BEFORE the session invitation
          is answered. When a session invitation involving ToIP originates
          from a gateway, this MAY be signaled to the user.
          The user MUST be informed of any change in modalities.
       6.2.7 Session progress information
          During a conversation that includes ToIP, status and session
          progress information MUST be provided in a textual form so users
          can perform all session control functions. That information MUST
          be equivalent to session progress information delivered in any
          other format, for example audio.
          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.
          There MUST be a clear indication, in a modality useful to the
          user, whenever a session is connected or disconnected. 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
          In summary, it SHOULD be possible to observe indicators about:
          - Incoming session
          - Availability of text, voice and video channels
          - Session progress
          - Incoming text
          - Any loss in incoming text
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          - Typed and transmitted text.
          For users who cannot use the audible alerter for incoming
          sessions, it is RECOMMENDED to include a tactile as well as a
          visual indicator.
       6.2.8 Session Negotiations
          The Session Description Protocol (SDP) used in SIP [3] provides
          the capabilities to indicate text as a medium in the session
          setup. RFC 4103 [5] uses the RTP payload type "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,
          SIPÆs offer/answer model [20] can also be used in conjunction with
          other capabilities including the use of a transcoding server for
          enhanced session negotiations [7,8,9].
       6.2.9 Answering
          Systems SHOULD provide a best-effort approach to answering
          invitations for session set-up and users SHOULD be informed when
          the session is accepted by the other party. On all systems that
          both inform users of session status and support ToIP, this
          information MUST be available in textual form and MAY also be
          provided in other 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
          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.
          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.
       6.2.10 Actions During a Session
          Certain actions need to be performed during ToIP conversation:
          a. 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.
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          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
          c. To enable the use of international character sets, the
             transmission format for text conversation SHALL be UTF-8 [12],
             in accordance with ITU-T T.140.
          d. If text is detected to be missing after transmission, there
             SHOULD be a "text loss" indication in the text as specified in
             T.140 Addendum 1 [4].
          e. When the display of text conversation is included in the design
             of the end user equipment, the display of the dialogue SHOULD
             be made so that it is easy to differentiate the text belonging
             to each party in the conversation.
      Text Transport
          ToIP uses RTP as the default transport protocol for the
          transmission of real-time text via the medium "text/t140" as
          specified in RFC 4103 [5].
          The redundancy method of RFC 4103 [5] SHOULD be used to
          significantly increase the reliability of the text transmission. A
          redundancy level using 2 generations gives very reliable results
          and is therefore RECOMMENDED.
          Text capability MUST be announced in SDP by a declaration similar
          to 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
          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 [5] transport MUST be used as both the
          default and the fallback transport.
      Handling Text and other Media.
          A call is one or more related sessions. The following requirements
          apply to media handling during a call:
          a. When used between User Agents designed for ToIP, it SHALL be
             possible to send and receive text simultaneously.
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          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
          c. It SHOULD be possible to know during a call that ToIP is
             available, even if it is not invoked at call setup (e.g. when
             only voice and/or video is used initially). To disable this,
             the user MUST disable the use of ToIP. This is possible during
             registration at the SIP registrar.
       6.2.11 Additional session control
          Systems that support additional session control features, for
          example call waiting, forwarding, hold etc on voice sessions, MUST
          offer this functionality for text sessions.
       6.2.12 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: a UTF8 text file in XML format [11] including timestamps,
          party names (or addresses) and the text conversation.
       6.3 Conference Session Requirements
          The conference session requirements deal with multipoint
          conferencing sessions where there will be one or more ToIP capable
          devices and/or other end user devices where the total number of
          end user devices will be at least three.
          It SHOULD be possible to use the text medium in conference
          sessions in a similar way to how audio is handled and 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 Real-time Editing and User Alerting
          ToIP SHOULD handle characters such as new line, erasure and
          alerting during a session as specified in ITU-T T.140.
       6.5 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 text medium MUST
          be equivalent to the emergency service provided to users utilising
          speech or other media.
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       6.6 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.
       6.7 Firewalls and NATs
          ToIP uses the same signaling 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. 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 the interoperability
          requirements, especially for 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
          (due to a hearing impairment), or who can hear but must send text
          (due to a speech impairment).
          Session setup through gateways to other networks MAY require the
          use of specially formatted addresses or other mechanisms for
          invoking those gateways.
          Different data rates of different protocols MAY require text
          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.
       7.2 ToIP and PSTN/ISDN Text-Telephony Interworking
          On PSTN networks, transmission of interactive text takes place
          using a variety of codings and modulations, including ITU-T V.21
          [II], Baudot [III], DTMF, V.23 [IV] 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.
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          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.
          V.18 MUST be supported on the PSTN side of a PSTN-ToIP gateway.
          PSTN-ToIP gateways MUST allow alternating use of text and voice if
          the PSTN textphone involved at the PSTN side of the session
          supports this. (This mode is often called VCO/HCO).
          Calling party identification information, such as CLI, MUST be
          passed by gateways and converted to an approapriate form if
       7.3 ToIP and Cellular Wireless ToIP
          ToIP MAY be supported over the cellular wireless packet switched
          service. It interfaces to the Internet.
          A text gateway with cellular wireless packet switched services
          MUST be able to route text calls to emergency service providers
          when any of the recognized emergency numbers that support text
          communication for the country.
       7.4 Instant Messaging Support
          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, gateways MUST transcode between the 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 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.
          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.
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          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 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 are the
          same for Instant Messaging (SIMPLE) and ToIP.
          Devices that implement Instant Messaging SHOULD implement ToIP as
          described in this document so that a more complete text
          communication service can be provided.
       7.5 Common Text Gateway Functions
          Text gateways MUST allow for the differences that result from
          different text protocols. The protocols to be supported will
          depend on the service requirements of the Gateway.
       7.5.1 Protocol support
          Text gateways MUST use the ITU-T V.18 [10] standard at the PSTN
          side. A text gateway MUST act as a SIP User Agent on the IP side
          and support RFC4103 text transport.
       7.5.2 Relay buffer storage
          When text gateway functions are invoked, there will be a need for
          intermediate storage of characters before transmission to a device
          receiving text slower than the transmitting 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 for up to 4 minutes (i.e. less than 3k
          Interoperation of half-duplex and full-duplex protocols 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
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          at the "protocol" level (where an indication must be sent back to
          the Gateway).
       7.5.3 Emergency calls through gateways
          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 e.g. "911" in USA and "112" in Europe. Routing text calls
          to emergency services MAY require the use of a transcoding
       7.5.4 Text Gateway Invocation
          ToIP interworking 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 generally only able to receive voice
          and streaming text, so the text gateway will only allow ToIP and
          Examples of possible scenarios for invocation of the text gateway
          a. PSTN textphone users dial a prefix number before dialing out.
          b. Separate text subscriptions, linked to the phone number or
             terminal identifier/ IP address.
          c. Text capability indicators.
          d. Text preference indicator.
          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 here
          h. Text gateways with its own telephone number and/or SIP address.
             (This requires user interaction with the text gateway to place
             a call).
          i. ENUM address analysis and number plan
          j. Number or address analysis leads to a gateway for all PSTN
       7.6 Home Gateways or Analog Terminal Adapters
          Analog terminal adapters (ATAs) using SIP based IP communication
          and RJ-11 connectors for connecting traditional PSTN devices
          SHOULD enable connection of legacy PSTN text telephones [16].
          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 sessions.
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          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.
          NOTE: It is preferred that such analog terminal 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 character loss in the textphone
          conversation or even not allowing the textphones to connect to
          each other.
       7.7 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 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 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.
          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,
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          where this character exists. For legacy systems where the
          character ' does not exist, the . ( full stop ) character SHOULD
          be used instead.
       7.9 Relay Services
          The relay service acts as an intermediary between two or more
          callers using different media or different media encoding schemes.
       7.9.1 Basic function of the relay service
          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 other
          relay services exist, like video relay services transcoding speech
          to sign language and vice versa where the signing is communicated
          using video.
       7.9.2 Invocation of relay services
          It is RECOMMENDED that ToIP implementations make the invocation
          and use of relay services as easy as possible. It 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 [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 session.
          When setting up a session, 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 session to a preferred relay
          service even if the user invokes the session from another region
          or network than that usually used.
          A number of requirements, motivations and implementation
          guidelines for relay service invocation can be found in RFC 3351
       8. Security Considerations
          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
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          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.
       9. 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
          Barry Dingle
          ACIF, 32 Walker Street
          North Sydney, NSW 2060 Australia
          Tel +61 (0)2 9959 9111
          Mob +61 (0)41 911 7578
          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
          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
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          Radhika R. Roy
          3465-B Box Hill Corporate Center Drive
          Abingdon, MD 21009
          Tel: 443 402 9041
          Henry Sinnreich

          115 Broadhollow Rd
          Suite 225
          Melville, NY 11747
          Tel: +1.631.961.8950
          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
          Centre for R & D on sensory and communication disabilities.
          Theerestraat 42
          5271 GD Sint-Michielsgestel
          The Netherlands.
       10. References
       10.1 Normative references
          1. S. Bradner, "Intellectual Property Rights in IETF Technology
          ", BCP 79, RFC 3979, IETF, March 2005.
          2. S. Bradner, "Key words for use in RFCs to Indicate Requirement
          Levels", BCP 14, RFC 2119, IETF, 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).
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          5. G. Hellstrom, "RTP Payload for Text Conversation", RFC 4103,
          IETF, 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,
          IETF, June 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
          12. Yergeau, F., "UTF-8, a transformation format of ISO 10646",
          RFC 2279, IETF, January 1998.
          13. TIA/EIA/IS-823-A  "TTY/TDD Extension to TIA/EIA-136-410
          Enhanced Full Rate Speech Codec (must used in conjunction with
          14. TIA/EIA/IS-127-2 "Enhanced Variable Rate Codec, Speech Service
          Option 3 for Wideband Spread Spectrum Digital Systems. Addendum
          15. 3GPP TS26.226  "Cellular Text Telephone Modem Description"
          16. H. Sinnreich, S. Lass,  and C. Stredicke, "SIP Telephony
          Device Requirements and Configuration," IETF, June 2005 - Work in
          17.  Baugher, McGrew, Carrara, Naslund, Norrman, "The Secure Real
          Time Transport Protocol (SRTP)", RFC 3711, IETF, March 2004.
          18. "IP Multimedia default codecs". 3GPP TS 26.235
          19. Charlton, Gasson, Gybels, Spanner, van Wijk, "User
          Requirements for the Session Initiation Protocol (SIP) in Support
       A. van Wijk, et al.     Expires 6 March 2006      [Page 26 of 28]
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          of Deaf, Hard of Hearing and Speech-impaired Individuals", RFC
          3351, IETF, August 2002.
          20. J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with the
          Session Description Protocol (SDP)", RFC 3624, IETF, June 2002.
          21. ITU-T Recommendation F.700,"Framework Recommendation for
          Multimedia Services", November 2000.
          22. H. Schulzrinne, S.Casner, R. Frederick, V. Jacobsone, "RTP: A
          Transport Protocol for Real-Time Applications", RFC 3550, IETF,
          July 2003.
          23. ITU-T Recommendation F.703,"Multimedia Conversational
          Services", November 2000.
          24. J. Rosenberg, H. Schulzrinne, P. Kyzivat, "Indicating User
          Agent Capabilities in the Session Initiation Protocol (SIP)", RFC
          3840, IETF, August 2004
       10.2 Informative references
          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
          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. 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.)
          IV. International Telecommunication Union (ITU), "600/1200-baud
          modem standardized for use in the general switched telephone
          network. ITU-T Recommendation V.23, November 1988.
       Intellectual Property Statement
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       Internet-Draft Requirements for real time text using SIP    Sept 2005
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