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       Internet Engineering Task Force                 SIPPING WG
       Internet Draft                                  G. Hellström (editor)
       Document: <draft-manyfolks-sipping-ToIP-01.txt> R. R. Roy (editor)
       Feb 2004                                        A. van Wijk (editor)
       Expires: August 2004                            Omnitor, AT&T,
       Informational                                   Viataal
       
       
        Framework of requirements for real-time text conversation using SIP.
       
       
       Status of this Memo
       
          This document is an Internet-Draft and is in full conformance with
          all provisions of Section 10 of RFC2026 [1].
          Internet-Drafts are working documents of the Internet Engineering
          Task Force (IETF), its areas, and its working groups. Note that
          other groups may also distribute working documents as Internet-
          Drafts.
       
          Internet-Drafts are draft documents valid for a maximum of six
          months and may be updated, replaced, or obsoleted by other
          documents at any time. It is inappropriate to use Internet-Drafts
          as reference material or to cite them other than as "work in
          progress."
       
          The list of current Internet-Drafts can be accessed at
          http://www.ietf.org/ietf/1id-abstracts.txt.
          The list of Internet-Draft Shadow Directories can be accessed at
          http://www.ietf.org/shadow.html.
       
       Abstract
       
          This document provides the framework of requirements for text
          conversation with real time character-by-character interactive
          flow over the IP network using the Session Initiation Protocol.
          The requirements for general real-time text-over-IP telephony,
          point-to point and conference calls, transcoding, relay services,
          user mobility, interworking between text-over-IP telephony and
          existing text-telephony, and some special features including
          instant messaging have been described.
       
       
       Table of Contents
       
       1. Introduction                                                 3
       2. Scope                                                        3
       3. Terminology                                                  3
       4. Definitions                                                  4
       5. Background and General Requirements                          5
       6. Features in Real-time Text-over-IP                           5
       7. Real-Time Multimedia Conversational Sessions using SIP       6
       8. General Requirements for Real-Time Text-over-IP using SIP    8
       
       Hellström, Roy, van Wijk                                [Page 1 of 34]
       
       draft-manyfolks-sipping-ToIP-01.txt                      February 2004
       
       8.1 Pre-Call Requirements                                       8
       8.2 Basic Point-to-Point Call Requirements                      9
       8.2.1 General Requirements                                      9
       8.2.2 Session Setup                                             9
       8.2.3 Addressing                                                10
       8.2.4 Alerting                                                  10
       8.2.5 Call Negotiations                                         10
       8.2.6 Answering                                                 11
       8.2.7 Session progress and status presentation                  11
       8.2.8 Actions During Calls                                      12
       8.2.9 Additional session control                                13
       8.2.10 File storage                                             14
       8.3 Conference Call Requirements                                14
       8.4 Transport                                                   14
       8.5 Character Set                                               14
       8.6 Transcoding                                                 15
       8.7 Relay Services                                              15
       8.8 Emergency services                                          16
       8.9 User Mobility                                               16
       8.10 Confidentiality and Security                               16
       8.11 Call Flows                                                 17
       8.11.1 Call Scenarios                                           17
       8.11.2 Point-to-Point Call Flows                                18
       8.11.3 Conference Call Flows                                    18
       9. Interworking Requirements for Text-over-IP                   19
       9.1 Real-Time Text-over-IP Interworking Gateway Services        19
       9.2 Text-over-IP and PSTN/ISDN Text-Telephony                   19
       9.3 Text-over-IP and Cellular Wireless circuit switched Text-
       Telephony                                                       20
       9.3.1 “No-gain”                                                 20
       9.3.2 Cellular Text Telephone Modem (CTM)                       20
       9.3.3 “Baudot mode”                                             21
       9.3.4 Data channel mode                                         21
       9.3.5 Common Text Gateway Functions                             21
       9.4 Text-over-IP and Cellular Wireless Text-over-IP             21
       9.5 Instant Messaging Support                                   21
       9.6 IP Telephony with Traditional RJ-11 Interfaces              22
       9.7 Interworking Call Flows                                     23
       9.8 Multi-functional gateways                                   24
       9.9 Gateway Discovery                                           24
       9.10 Text Gateway in the call Scenarios                         25
       9.10.1 IP terminal calling an analogue textphone (PSTN)         25
       9.10.2 IP terminal calling a mobile text telephone (CTM)        25
       9.10.3 IP terminal calling a mobile telephone (GPRS based)      25
       9.10.4 IP terminal calling a mobile telephone(UMTS)             26
       9.10.5 Analogue textphone (PSTN) user calling an IP terminal    26
       9.10.6 Mobile text telephone (CTM) user calling an IP terminal  26
       9.10.7 Mobile telephone user (GPRS) calling an IP terminal      26
       9.10.8 Mobile telephone (UMTS) user calling an IP terminal      26
       9.10.9 Voice over DSL user using an analogue text telephone.    27
       9.10.10 VoIP user via a building telephone switch (at an apartment
       building) owning an analogue text telephone.                    27
       
       
       Hellström, Roy, van Wijk                                [Page 2 of 34]
       
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       9.10.11 VoIP user via a gateway/box connected to his/her own
       Broadband connection owning an analogue text telephone.         27
       10. Terminal Features                                           27
       10.1 Text input                                                 28
       10.2 Text presentation                                          28
       10.3 Call control                                               29
       10.4 Device control                                             30
       10.5 Alerting                                                   30
       10.6 External interfaces                                        30
       10.7 Power                                                      31
       11. Security Considerations                                     31
       12. Authors’ Addresses                                          31
       13. Full Copyright Statement                                    32
       14. References                                                  33
       14.1 Normative                                                  33
       14.2 Informative                                                34
       
       
       1. Introduction
       
          Text-over-IP (ToIP) is becoming popular as a part of total
          conversation among a range of users although this medium of
          communications may be the most convenient to certain categories of
          people (e.g., deaf, hard of hearing and speech-impaired
          individuals). The Session Initiation Protocol (SIP) has become the
          protocol of choice for control of Multimedia IP telephony and
          Voice-over-IP (VoIP) communications. Naturally, it has become
          essential to define the requirements for how ToIP can be used with
          SIP to allow text conversations as an equivalent to voice. This
          document defines the framework of requirements for using ToIP,
          either by itself or as a part of total conversation using SIP for
          session control.
       
       2. Scope
       
          The primary scope of this document is to define the requirements
          for using ToIP with SIP, either stand-alone or as a part of a
          total conversation approach. In general, the scope of the
          requirements is:
       
          a. Features in Real-Time ToIP
          b. Real-time Multimedia Conversational Sessions using SIP
          c. General Requirements for Real-Time ToIP using SIP
          d. Interworking Requirements for ToIP
          e. Text gateways in the different networks
       
          The subsequent sections describe those requirements in detail.
       
       3. Terminology
       
          The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
          NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL"
       
       
       Hellström, Roy, van Wijk                                [Page 3 of 34]
       
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          in this document are to be interpreted as described in RFC 2119
          [2].
       
       4. Definitions
       
          Full duplex – user information is sent independently in both
          directions.
       
          Half duplex – user information 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 user information.
       
          TTY – name for text telephone, often used in USA, see textphone.
       
          Textphone –text telephone. A terminal device that allow end-to-end
          real time 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 telephony – 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 (human
          or automated) that transcodes one stream into another.
       
          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.
       
          Text gateway – A multi functionalgateway that sits at the border
          of a network able to transcode RFC 2793 Interactive text (ToIP)
          into a different text medium and vice versa. E.g. ToIP into Boudot
          and vice versa in the PSTN.
       
          Acronyms:
       
          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
       
       Hellström, Roy, van Wijk                                [Page 4 of 34]
       
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          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. Background and General Requirements
       
          The main purpose of this document is to provide a set of
          requirements for real-time text conversation over the IP network
          using the Session Initiation Protocol (SIP) [3]. The overall
          requirements described are such that the real-time text can be
          expressed as a part of the session description as a part of the
          total conversation like any other media. Participants can
          negotiate all media including real-time text conversation[4, 5].
          This is a highly desirable function for all IP telephony
          users,irrespective of whether the users are or are not deaf, hard
          of hearing, or 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 take place, therefore providing the same mode of
          interaction as voice telephony does. Store-and-forward systems
          like email or messaging on mobile networks or non-streaming
          systems like instant messaging are unable to provide that
          functionality.
       
          One particular application where real-time text is absolutely
          essential, is the use of relay services between conversational
          modes, like between text and voice.
       
          Direct text emergency service calls, where time and continuous
          connection are of the essence, is another essential application.
       
       6. Features in Real-time Text-over-IP
       
          While real-time Text-over-IP will be used for a wide variety of
          services, an important field of application will be to provide a
          text equivalent to voice conversation, in particular for deaf,
          hard of hearing and speech-impaired users.
          As such, it is crucial that the conversational nature of this
          service is maintained. Text based communications exist in a
          variety of forms, some non-conversational (SMS, text paging, E-
          mail, newsgroups, message boards, etc.), others conversational
          (TTY/TDD, Textphone, etc).
       
          Real-time Text-over-IP will sometimes be used in conjunction with
          a relay service [I] to allow text users to communicate with voice
          users. With relay services, it is crucial that text characters are
       
       Hellström, Roy, van Wijk                                [Page 5 of 34]
       
       draft-manyfolks-sipping-ToIP-01.txt                      February 2004
       
          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.
       
          In order to make Real-Time Text-over-IP the equivalent of what
          voice is to hearing people, it needs to offer equivalent features
          in terms of conversation as voice communications provides to
          hearing people. To achieve that, real-time Text-over-IP MUST:
           a. Offer Real-Time presentation of the conversation. This means
          that text MUST be sent as soon as available, or with very small
          delays. The delay MUST not be longer than 500 milliseconds,
           b. Provide simultaneous transmission in both directions,
           c. Except for the case of interworking with other networks and
          protocols (e.g. TTY on PSTN) allow users to interrupt/barge in at
          any time in the conversation.
           d. Except for the case of interworking with other networks and
          protocols, Real-Time Text-over-IP MUST support a transmission rate
          of at least 30 characters/second.
           e. Support sending redundant data as described in RFC 2793 [5].
           f. Be possible to merge with video transmission.
       
          The end-to-end delay in transmission MUST be less than 2000
          milliseconds.
       
          Many users will want to use multiple modes of communication during
          the conversation, either at the same time or by switching between
          modes e.g. between real-time Text-over-IP and voice. Native real-
          time Text-over-IP systems MUST support at least the alternate use
          of modalities and MAY support simultaneous use of modalities.
       
          When communicating via a gateway to other networks and protocols,
          the system MUST completely support the functionality for
          alternating or simultaneous modalities as offered by the gateway.
          When voice is supported on the terminal, the terminal MUST provide
          volume control.
       
       7. Real-Time Multimedia Conversational Sessions using SIP
       
          The Session Initiation Protocol (SIP) [3] provides mechanisms for
          creating, modifying, and terminating sessions for real-time
          conversation with one or more participants using any combination
          of media: Text, Video and Audio. However, participants are allowed
          to negotiate on a set of compatible media types (e.g., Text,
          Video, Audio) with session descriptions used in SIP invitations.
       
          The standardized T.140 real-time text conversation [4], in
          addition to audio and video communications, will be valuable
          services to many. Real-time text can be expressed as a part of the
          session description in SIP and will be a useful subset of the
          Total Conversation (e.g., Real-time text, Video and Audio).
       
       
       
       Hellström, Roy, van Wijk                                [Page 6 of 34]
       
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          This specification describes the 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.
       
          The session establishment using SIP defines procedures for how
          T.140 text conversation can be supported using a RTP payload
          defined in RFC 2793 [5]. The performance characteristics of T.140
          will be determined using RTCP.
       
          The session will not only define procedures between the SIP
          devices having text conversation capability, but will also define
          how sessions in SIP can be established between the text
          conversation and audio/video/text capable devices transparently.
       
          If there is any incompatibility between the terminals, e.g. T.140
          only and audio-only terminals, the necessary transcoding services
          will need to be invoked. This important service feature invites a
          variety of rich capabilities in the transcoding server. For
          example, speech-to-text (STT), text-to-speech (TTS), text bridging
          after conversion from speech, audio bridging after conversion from
          text, and other services can also be provided by the transcoding
          and/or translation server. The session description protocol (SDP)
          [6] used in SIP to describe the session also needs to be capable
          of expressing these attributes of the session (e.g., uniqueness in
          media mapping for conversion from one media to another for each
          communicating party).
       
          Real-time texts can also be presented in conjunction with video.
       
          Alerting for T.140 terminals needs to be provided. Users may set
          up text conversation sessions using SIP from any location. In
          addition, user privacy and security MUST be provided for text
          conversation sessions at least equal to that for voice.
       
          The transcoding/translation services can be invoked in SIP using
          different 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, the interworking between T.140 text
          conversation and text telephony conversation [10] is needed.
       
          The general requirements for real-time text conversation using SIP
          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. Negotiation procedures for device capabilities
          d. Discovery and invocation of transcoding/translation services
          between the media in the call
       
       
       Hellström, Roy, van Wijk                                [Page 7 of 34]
       
       draft-manyfolks-sipping-ToIP-01.txt                      February 2004
       
          e. Different session establishment models for
          transcoding/translation services invocation: Third party call
          control and Conference bridge model
          f. 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
          g. Media bridging services for T.140 real-time text, audio, and
          video for multipoint communications
          h. Transparent session setup, modification, and teardown between
          text conversation capable and voice/video capable devices
          i. Conversations to be carried out using T.140-over-RTP and RTCP
          will provide performance report for T.140
          j. Altering capability using text conversation during the session
          establishment
          k. T.140 real-time text presentation mixing with voice and video
          l. T.140 real-time text conversation sessions using SIP, allowing
          users to move from one place to another
          m. Users’ privacy and security for sessions setup, modification,
          and teardown as well as for media transfer
          n. Interoperability between T.140 conversations and text telephony
       
       8. General Requirements for Real-Time Text-over-IP using SIP
       
          The communications environments for ToIP using SIP to set up the
          conversation in real-time may vary from a simple point-to-point
          call to multipoint calls in addition to the fact that ToIP can be
          used in combination with other media like audio and video. In
          order to establish the session in real-time, the communicating
          parties SHOULD be provided with experiences like those of normal
          telephony call setup. There may also 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 addressing.
       
          Similarly, there are requirements that need to be satisfied during
          call set up when another media is preferred by a user. For
          instance, some users may prefer to use audio while others want to
          use text as their preferred choice of conversational mode. In this
          case, transcoding services will need to be invoked 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 those requirements in great
          detail.
       
       8.1 Pre-Call Requirements
       
          The desire of the users for using ToIP as a medium of
          communications can be expressed during registration time. Two
          situations need to be considered in the pre-call setup
          environment:
       
       
       Hellström, Roy, van Wijk                                [Page 8 of 34]
       
       draft-manyfolks-sipping-ToIP-01.txt                      February 2004
       
          a. User Preferences: It MUST be possible for a user to indicate a
          preference for ToIP by registering that preference in a SIP
          server. If the user is called by other party, preferences can be
          invoked by the SIP server to accept or reject the call based on
          the rules defined by the user. If the rules require that a
          transcoding server is needed, the call can be re-directed or
          handled accordingly.
       
          b. Server to support User Preferences: SIP servers MUST have the
          capability to act on users preferences for ToIP, based on the
          users’ preferences defined during the pre-call setup registration
          time.
       
       8.2 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 the preferred medium for conversation using SIP in the session
          setup.
       
       8.2.1 General Requirements
       
          The general requirements are that ToIP will be chosen from the
          available media as the preferred means of communication for the
          session. However, there may be a need to invoke some underlying
          capabilities in some cases, for example, a transcoding server may
          be invoked if one of the users want to use a communication medium
          other than ToIP.
          The following entities MAY need to be involved to facilitate the
          session establishment using ToIP as another medium:
       
          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).
       
       8.2.2 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 real-time Text-over-IP. For
          instance, the conversation might be established using voice and
          the user could elect to switch to text, or add text, during the
          conversation. Systems supporting real-time Text-over-IP MUST allow
       
       
       Hellström, Roy, van Wijk                                [Page 9 of 34]
       
       draft-manyfolks-sipping-ToIP-01.txt                      February 2004
       
          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 voice, 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. and 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 prefixes or the use of specially formatted
          URLs.
          This MUST be supported by the terminal.
       
       8.2.3 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 transforming or translating service MUST
          NOT require user registration in any specific SIP registrar.
       
       8.2.4 Alerting
       
          Systems supporting real-time Text-over-IP MUST have an alerting
          method (e.g., for incoming calls and messages) 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 on the user equipment 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 real-time
          Text-over-IP originates from a gateway, this MAY be signalled to
          the user.
       
       8.2.5 Call Negotiations
       
       Hellström, Roy, van Wijk                               [Page 10 of 34]
       
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          The Session Description Protocol (SDP) used in SIP [3] provides
          the capabilities to indicate ToIP as a media for the call setup.
          RFC 2793 [5] provides the RTP payload type 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, SIP’s
          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].
       
       8.2.6 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 real-
          time Text-over-IP, this information MUST be available in text, and
          may be provided in other visual media.
       
       8.2.6.1 Auto-Answer
       
          Systems for real-time Text-over-IP MAY support an auto-answer
          function, equivalent to answering machines on telephony networks.
          If an auto-answer function is supported, it MUST support at least
          160 characters for the recorded message. It MUST support  incoming
          text message storage of a minimum of 16000 characters, although
          systems MAY support much larger storage.
       
          When the auto-answer function is activated, user alerting MUST
          still take place. The user MUST be allowed to monitor the auto-
          answer progress and MUST be allowed to intervene during any stage
          of the auto-answer and take control of the session.
       
       8.2.7 Session progress and status presentation
       
          During a conversation that includes real-time Text-over-IP, 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 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 MUST use simple language as much as
          possible so that it can be understood by as many users as
          possible.
          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 and disconnected. The user
       
       Hellström, Roy, van Wijk                               [Page 11 of 34]
       
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          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.
       
       8.2.8 Actions During Calls
       
          Certain actions need to be performed for the ToIP conversation
          during the call and these actions are describe briefly as follows:
       
          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
          milliseconds.
          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. If text is detected to be missing after transmission, there
          SHALL be an indication in the text marking the loss.
          g. When used from a terminal designed for PSTN text telephony, or
          in interworking with such a terminal, ToIP shall enable
          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 USA).
          h. The transmission of the text conversation SHALL be made
          according to an internationally suitable character set and control
          protocol for text conversation as specified in ITU-T T.140.
          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
          conversation.
       
       8.2.8.1 Text and other Media Handling Between ToIP Devices
       
          The native ToIP devices do not need transcoding from speech to
          text and can communicate directly.
       
          I. When used between terminals designed for native 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.
       
       
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          II. When used between terminals designed for native ToIP, it SHALL
          be possible to send and receive text simultaneously.
       
       8.2.8.2 General Actions
       
          a. It SHALL be possible to establish a session with text
          capabilities enabled at the beginning of a  Call. Note: a call is
          in this document defined as one or more sessions).
          b. It SHALL be possible to place a call without text capabilities,
          and to add text capabilities later in the call.
          c. It SHALL be possible to transfer text at at least 30 characters
          per second
          d. It SHALL be possible to talk and listen simultaneously with
          typing and reading.
       
       8.2.8.3 Call Action with Native ToIP Devices
       
          a. It SHALL be possible to answer a callwith text capabilities
          enabled.
          b. It SHOULD be possible to use video simultaneously with the
          other media in the call.
          c. It SHALL  be possible to answer a callin voice or video without
          text enabled, and add text later in the call.
          d. It SHALL be possible to disconnect the call.
          e. It SHOULD be possible to control IVR (Interactive Voice
          Response) services from a numeric keypad.
          f. It SHOULD be possible to control ITR ( Interactive Text
          Response) services from the alphanumeric keyboard.
          g. It SHOULD be possible to invoke multi-party calls.
          h. It SHALL be possible to transfer the call.
          i. It SHOULD be possible to use text characters (numbers) instead
          of DTMF tones (numbers) in interactions where the person is using
          a keyboard to interact with a service and the service asks for a
          number.
       
       8.2.8.4 Audio/Visual/Tactile Indicators
       
          It SHALL 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.
       
       8.2.9 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 real-time Text-over-IP
          functions. In addition, all these features MUST be controllable by
          text users at any time, in an equivalent way as for other users.
       
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          It SHOULD be possible to use text characters (numbers) instead of
          DTMF tones (numbers) in interactions where the person is using a
          keyboard to interact with a service and the service asks for a
          number.
       
       8.2.10 File storage
       
          Systems that support real-time Text-over-IP MAY save the text
          conversation to a file. This SHOULD be done using a standard file
          format. It is recommended to use an xhtml [11] format.
       
       8.3 Conference Call Requirements
       
          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.
       
       8.4 Transport
       
          ToIP SHALL use RTP as the default transport protocol for
          transmission of real-time text as specified in RFC 2793 [5].
          Signaling and other media will use the transport protocol
          specified in SIP [3] and/or their revised versions as specified in
          standards.
       
          The redundancy method of RFC 2198 SHOULD be used for making text
          transmission reliable with transmission of three generations.
       
          Text capability SHOULD 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
       
          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 optimised.
       
          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 2793 transport is used as the
          defaut fallback solution.
       
       8.5 Character Set
       
       
       
       
       Hellström, Roy, van Wijk                               [Page 14 of 34]
       
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          a. Real-Time Text-over-IP protocols MUST use UTF-8 encoding as
          specified in ITU-T T.140 [12]. A number of characters used in
          traditional text telephony have special meanings.
          b. Real-time Text-over-IP SHALL handle characers with editing
          effect such as new line, erasure and alerting during session as
          specified in ITU-T T.140.
       
       8.6 Transcoding
       
          Transcoding of text may need to take place in gateways between
          ToIP and other forms of text conversation. ToIP make use of 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 these character sets and T.140 in
          gateways, special consideration MUST be paid 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 SHOULD be possible to select by the user per call, 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 SHALL be translated to be represented by the '
          (apostrophe) character in legacy text telephone systems where this
          character exists. For legacy systems where the character ' does
          not exist, the character . ( full stop ) SHALL be used instead.
       
       8.7 Relay Services
       
          The relay service acts as an intermediary between 2 or more
          callers.
          The basic 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 do not exclude video relay services for e.g.,
          speech to sign language and vice versa and other possible relay
          services. It will be possible to use ToIP simultaneously with
          other relay services if desired.
       
          It is very important for the users that a relay session is invoked
          as transparently as possible. It SHOULD happen automatically when
          the call is being set-up or by a simple user action. 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.
       
          Adding or removing a relay service MUST be possible without
          disrupting the current call.
       
       
       
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          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 including the sign language being used.
       
          The user MUST be provided with a method to indicate which service
          is desired.
       
          It MUST be possible to identify ToIP sessions as emergency
          sessions.
       
          The relay service operator MUST be able to process such a session
          correctly and quickly.
       
          a. The relay service operator’s network must give priority to this
          incoming call.
          b. The relay service operator MUST forward this session if they
          are unable to process it to an alternative emergency relay
          operator.
          c. The relay service MUST label the transcoded stream as an
          emergency call (in case of text to speech and/or vice versa).
          d. The relay service MUST provide all session information to the
          emergency centre (e.g., location information of the caller if
          available).
       
          Relay services must be available all the time, even if the users
          are roaming.
       
       8.8 Emergency services
       
          a. It SHALL be possible to support emergency service calls with
          text only or simultaneously with voice.
          b. All session information that accompanies a voice session to the
          emergency centre, shall also be provided to the emergency centre
          if it is a ToIP session.(e.g, phone number and location
          information of the user placing the emergency call).
          c. A text over IP stream must be labelled as an emergency stream
          to ensure that the emergency service center is able to receive
          this call.
       
       8.9 User Mobility
       
          ToIP terminals SHALL use the same mechanisms as other terminals to
          resolve mobility issues. It is RECOMMENDED to use a SIP-adress for
          the users, resolved by a SIP REGISTRAR, to enable basic user
          mobility. Further mechanisms are defined for the 3G IP multimedia
          systems.
       
       8.10 Confidentiality and Security
       
          Users’ 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
       
       Hellström, Roy, van Wijk                               [Page 16 of 34]
       
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          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].
       
          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.
       
       8.11 Call Flows
       
          ToIP is a way of establishing the real-time conversation. Call
          flow for ToIP SHOULD be as similar to audio and video session
          establishment. For example, ToIP services MAY be invoked in the
          following situations (among others):
       
          - 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
          purposes)
          - Overcoming of language barriers through speech translation
          and/or transcoding services
          - Not hearing well or at all (e.g., hearing loss due to aging,
          heard of hearing, deaf)
       
          NOTE: In many of the above scenarios, text may accompany speech in
          a caption like fashion.  This would occur for individuals who are
          hard of hearing and also for mixed calls with a hearing and deaf
          person listening to the call.
       
          All call flows either for the point-to-point or for the multipoint
          need to consider that ToIP services may be invoked for many
          different reasons by users as explained. When the
          transcoding/translation services are needed, call flows will be
          shown for both session establishment models: Third-party call
          control model and Conferencing bridge model.
       
       8.11.1 Call Scenarios
       
          There are 2 different terminal types possible:
       
          1. The terminal itself has the intelligence to initiate a relay
          service for incoming and outgoing calls (based on address book,
          user preferences programmed on the terminal etc. This terminal can
          be used in a conference bridge call as well as a third party
          control call.
       
          2. Dumb terminals, so that the relay service server actually
          initiates the correct call handling (the dumb terminal can only
       
       Hellström, Roy, van Wijk                               [Page 17 of 34]
       
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          REFER the call to the relay center, which then sets up the call
          using the conference bridge flow.).
       
          The following call scenarios are shown:
       
          - Communications between two ToIP/Multimedia capable, end user
          devices using the same language.
          - Communications between ToIP capable, end user devices using
          translation services to provide language translation.
          - Communications between ToIP/Multimedia capable and Audio (non-
          ToIP) capable end user devices.
          - Communications between ToIP/Multimedia and/or Audio (non-
          ToIP)/Multimedia end user devices maintaining privacy.
       
       8.11.2 Point-to-Point Call Flows
       
          The point-to-point calls will contain at least one or both
          ToIP/Multimedia devices in setting up the session. The detail call
          flows need to be provided in the following scenarios:
       
          - ToIP/Multimedia devices that use the same language.
          - ToIP/Multimedia devices invoke translation services for using
          different languages.
             * Third-party call control model.
             * Conference bridge service model.
          - ToIP/Multimedia devices invoke translation services for using
          different languages maintaining privacy.
             * Third-party call control model.
             * Conference bridge service model.
          - ToIP/Multimedia device and Audio (non-ToIP)/Multimedia device
          invoking transcoding server.
             * Call initiated by Audio (non-ToIP)/Multimedia user
               - Third-party call control model.
               - Conference bridge service model.
             * Call initiated by ToIP user.
               - Third-party call control model.
               - Conference bridge service model.
          - ToIP/Multimedia device and Audio (non-ToIP)/Multimedia device
          invoking transcoding server maintaining privacy.
             * Call initiated by Audio (non-ToIP)/Multimedia user
               - Third-party call control model.
               - Conference bridge service model.
             * Call initiated by ToIP user.
               - Third-party call control model.
               - Conference bridge service model.
       
       8.11.3 Conference Call Flows
       
          Conference call flows only contain the multipoint communications
          scenarios, and only the centralized bridge model is considered.
          The following multipoint conference call flow scenarios will
          contain at least one more ToIP/Multimedia devices:
       
       
       Hellström, Roy, van Wijk                               [Page 18 of 34]
       
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          - ToIP/Multimedia devices that use the same language.
          - ToIP/Multimedia devices invoke translation services for using
          different languages.
          - ToIP/Multimedia devices invoke translation services for using
          different languages maintaining privacy.
          - ToIP/Multimedia device and Audio (non-ToIP)/Multimedia device
          invoking transcoding server.
             * Call initiated by Audio (non-ToIP)/Multimedia user.
             * Call initiated by ToIP/Multimedia user.
          - ToIP/Multimedia device and Audio (non-ToIP)/Multimedia device
          invoking transcoding server maintaining privacy.
             * Call initiated by Audio (non-ToIP)/Multimedia user.
             * Call initiated by ToIP/Multimedia user.
       
       9. Interworking Requirements for Text-over-IP
       
          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.
       
       9.1 Real-Time Text-over-IP 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. The 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.
       
       9.2 Text-over-IP 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 Real-time Text-
          over-IP 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.  2K is recommended. In addition,
          the gateway MUST provide a minimum throughput of at least 30
       
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          characters/second or the highest speed supported by the PSTN text
          telephony protocol side, whichever is the lowest.
       
          PSTN-Real-time Text-over-IP gateways MUST allow alternating use of
          text and voice.
       
          PSTN and ISDN to real-time Text-over-IP 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.
       
       9.3 Text-over-IP 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’
          solution.
       
          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 real-time Text-over-IP at
          the other side.
       
       9.3.1 “No-gain”
       
          The “No-gain” text telephone transporting technology uses
          specially modified EFR [15] and EVR [16] speech vocoders in both
          mobile terminals used provide a text telephony call. It provides
          full duplex operation and supports alternating voice and
          text.("VCO/HCO").
       
       9.3.2 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
          operation.
       
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       9.3.3 “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.
       
       9.3.4 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.
       
       9.3.5 Common Text Gateway Functions
       
          Text Gateways MUST support 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
          buffering.
       
          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
          Gateway).
       
          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 are called eg.
          ‘911’ in USA.
       
          A text gateway (MUST)/SHOULD act transparantly on the IP side. It
          acts then as a virtual end-point terminal.
       
       9.4 Text-over-IP and Cellular Wireless Text-over-IP
       
          Text-over-IP 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 into emergency service providers
          when any of the recognized emergency numbers that support text
          communication for the country are called.
       
       9.5 Instant Messaging Support
       
          Instant Messaging is used by many people to communicate using text
          via the Internet. Instant Messaging transfers blocks of text
       
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          rather than streaming as is used for real-time Text-over-IP. As
          such, it is not a replacement for real-time Text-over-IP and in
          particular does not meet the needs for real time conversations of
          deaf, hard of hearing and speech-impaired users. It is unsuitable
          for communications through a relay service [I]. The streaming
          character of real-time Text-over-IP provides  a better user
          experience and, when given the choice, users often prefer real-
          time Text-over-IP.
       
          However, since some users might only have Instant Messaging
          available, text gateways might be developed that allow
          interworking between Instant Messaging systems and real-time Text-
          over-IP 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 real-time Text-over-IP
          MUST concatenate individualcharacters originating at the real-time
          Text-over-IP side into blocks of text and:
       
          a. When the length of the concatenated message becomes longer than
          50 characters, the buffered text MUST be transmitted to the
          Instant Messaging side as soon as any non-alphanumerical character
          is received from the real-time Text-over-IP side.
       
          b. When a single carriage return, a single line feed, a carriage
          return/line feed pair or a line feed/carriage return pair is
          received from the real-time Text-over-IP side, the buffered
          characters up to that point, including the carriage return and/or
          line feed characters, MUST be transmitted to the Instant Messaging
          side.
       
          c. When the real-time Text-over-IP side has been idle for at least
          5 seconds, all buffered text up to that point MUST be transmitted
          to the Instant Messaging side.
       
          Many Instant Messaging protocols signal that a user is typing to
          the other party in the conversation. Text gateways between Instant
          Messaging and real-time Text-over-IP MAY provide this signaling to
          the Instant Messaging side when characters start being received,
          either at the beginning of the conversation.
       
          It is also possible to introduce the chat feature of certain
          Instant Messaging protocols. When the chat feature is selected,
          the IM client should use real-time text over IP. In this way, an
          IM client can also be used for real-time streaming text over IP.
       
       9.6 IP Telephony with Traditional RJ-11 Interfaces
       
          Analogue adapters using SIP based IP communication and RJ-11
          connectors for connecting traditional PSTN devices SHOULD enable
          connection of legacy PSTN text telephones [18]. These adapters
       
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          SHOULD contain V.18 modem functionality, voice handling
          functionality, and conversion functions to/from SIP based ToIP
          with T.140 transported in according to RFC 2793, in a similar way
          as it provides interoperability for voice calls. If a call is set
          up and RFC2793 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 RFC2793
          on board and thus act as a text gateway. Sending textphone signals
          over the voice channel is undesirable due posible filtering and
          compression between the 2 end-points. Which can result in dropping
          characters in the textphone conversation or even not allowing the
          textphones to connect with each other.
       
       9.7 Interworking Call Flows
       
          << this chapter will change depending on how chapter 9.10 works
          out>>
       
          The call flows in chapter 8.11 deal with end to end ToIP. These
          call flows do not change on the IP network when one end-point is
          actually a text gateway. The text gateway actually acts like a
          ToIP/Multimedia device. Separate call flows will show the
          interworking between the ToIP/Multimedia devices [4] over the IP
          network and the text telephony devices [10] over the PSTN/ISDN
          network using the IP-PSTN/ISDN interworking functional (IWF)
          entity. It is assumed that the IWF will provide ToIP and text
          telephony interworking in addition to other capabilities. Thus
          acting as a Text gateway.
       
          “The point-to-point call flows will contain at least one
          ToIP/Multimedia and one text telephony/multimedia (or POTS) device
          for the following cases:
       
          - ToIP/Multimedia device and text telephony/multimedia device that
          use the same/different language.
          - ToIP/Multimedia device and PSTN/ISDN-based POTS/Multimedia
          device.
       
          For multipoint conferencing calls, it is assumed that only the
          centralized conferencing will be considered, and the media bridge
          is supposed to be located somewhere in the SIP network. However,
          it is considered that the ToIP and text telephony interworking
          function will be located in the IWF.
       
          The multipoint conference call flows will contain at least one
          ToIP/Multimedia, at least one text telephony/multimedia device,
          and other devices where total number of devices will be three or
          more for the following cases:
       
       
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          - ToIP/Multimedia and text telephony/multimedia devices that use
          the same/different language.
          - ToIP/Multimedia devices, telephony/multimedia devices, and/or
          PSTN/ISDN-based POTS/Multimedia devices.”
       
       9.8 Multi-functional gateways
       
          The scenarios described in this document deal with single pairs of
          interworking protocols or services. However, in practice many of
          these interworking systems 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 real-time 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.
       
       
       9.9 Gateway Discovery
       
          To get a smooth invocation of the text gateways, where those
          gateways are transparant on the IP side, it requires a method how
          and when to invoke the text gateway. As described previously in
          this draft. The text gateways must act as the end-terminal. The
          capabilities of the text gateway will in that call 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. Thus the text gateway will only allow
          ToIP and, in case of VCO or HCO, audio.
       
          The PSTN devices or other non IP multimedia devices that require
          the text gateways to connect to the IP must be able to locate the
          text gateway. And ensure that the correct call capabilities of the
          non IP multimedia device is used by the text gateway.
       
          The following possible solutions for using the text gateway are:
       
          - PSTN Textphone users using a prefix before dialing out.
          - In band text dialogue. (???!!!)
          - separate text subscriptions, linked to the phone number or
          terminal identifier/ IP address.
          - text capability indicators.
          - text preference indicator.
          - listen for text activity in all calls.
          - call transfer request by the called user.
       
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          - placing a call via the web.
          - text gateways with its own telephone number and/or SIP address.
          (this requires user interaction with the text gateway to place a
          call).
          - ENUM.
          - etc
       
       9.10 Text Gateway in the call Scenarios
       
       9.10.1 IP terminal calling an analogue textphone (PSTN)
       
          The ToIP stream will be converted into an analogue text telephone
          protocol (using the voice channel) and vice versa by the text
          gateway.
       
          The PSTN knows it is a textphone call thanks to the SDP
          description (for example: m=text 11000 RTP/AVP 98 a=rtpmap:98
          t140/1000 for T.140 text on port 11000).
       
          The PSTN will also know that all those incoming calls are only for
          analogue textphones. Thus the speed of the text stream is adjusted
          to the selected analogue textphone protocol.
          If there is no analogue textphone on the called number, the call
          setup will be terminated by the text gateway.
       
          The text gateway can be implemented in 2 ways: The PSTN has its
          own text gateway (the IWF), or it redirects the media stream to
          the nearest IP-PSTN gateway with text transcoding abilities.
       
          Text gateway detection: In the SIP messages.
       
       9.10.2 IP terminal calling a mobile text telephone (CTM)
       
          The ToIP stream will be converted into CTM  and vice versa by the
          text gateway located in the network of the cellular/mobile
          operator. It is similar to the PSTN.
       
          Text gateway detection: In the SIP messages.
       
       9.10.3 IP terminal calling a mobile telephone (GPRS based)
       
          A text gateway located in the mobile network converts the incoming
          T.140/RTP stream into for example T.140 over TCP (T.140/TCP) or
          tunnels the T.140 stream over HTTP (T.140/HTTP). Or any other
          temporarily non standard solution necessary to connect the text
          gateway with the text telephone client on the mobile phone.
       
          This is necessary, since RTP over GPRS is not possible (especially
          on GPRS phones with Symbian OS).
          Note, those server-client solutions are ONLY acceptable for the
          GPRS and non RTP stack phones. It is encouraged to use T.140/RTP
          as soon as possible for all mobile phones.
       
       
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          Allowing UDP transport over the GPRS link will enable RFC2793 text
          over GPRS.
       
          Text gateway detection: In the SIP messages.
       
       9.10.4 IP terminal calling a mobile telephone(UMTS)
       
          No text gateway is required here since this will be end to end IP.
       
       9.10.5 Analogue textphone (PSTN) user calling an IP terminal
       
          The PSTN is unable to distinguish between an analogue voice call
          and an analogue textphone, both use the voice channel. The text
          gateway needed to transcode the analogue textphone protocol into
          T.140/RTP needs to be invoked.
       
          The easiest way for a PSTN to separate an incoming voice call into
          text telephony or normal voice is by using a prefix number for all
          incoming text telephone calls to the PSTN. For example , the text
          telephone user (e.g Boudot) places a call and enters a prefix e.g.
          600 and then continues with the original number. The PSTN will
          recognize all incoming 600 calls as an analogue textphone call and
          redirects the call to a text gateway (unless it is a number
          connecting the same PSTN).
       
          It is undesirable to allow a PSTN to transport all the analogue
          textphone tones/signals through a VoIP stream! (In band text
          dialogue).
       
          Text gateway detection: Prefix number for incoming textphone
          calls.
       
       9.10.6 Mobile text telephone (CTM) user calling an IP terminal
       
          The voice channel of the cellular network is used. The MSC is able
          to separate between the text call and voice only, it is just a
          matter of redirecting the voice channel to the text gateway.
       
          Text gateway detection: CTM signal detection.
       
       
       9.10.7 Mobile telephone user (GPRS) calling an IP terminal
       
          The text telephone client on the mobile telephone connects the
          text gateway located in the network. The text gateway transcodes
          the text stream into ToIP.
       
          Text gateway detection: pre-programmed in the mobile textphone
          client.
       
       9.10.8 Mobile telephone (UMTS) user calling an IP terminal
       
          No text gateway is required here since this will be end to end IP.
       
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       9.10.9 Voice over DSL user using an analogue text telephone.
       
          In Europe, Voice over DSL is introduced. It is likely that
          analogue text telephones just use the voice channel. The VoDSL
          gateway located in the network of the (A)DSL operator itself
          should connect with a text gateway as soon it turns into VoIP.
       
          Text gateway detection: prefix number similar to the PSTN.
       
       
       9.10.10 VoIP user via a building telephone switch (at an apartment
       building) owning an analogue text telephone.
       
          This is the case where only VoIP is possible and no other IP
          traffic between the telephone switch and the apartments. The
          question is if this will be implemented.
          The only solution would be a forced analogue text telephone
          protocol over the Voice channel, in band text dialogue . If that
          must happen. Then the telephone switch MUST convert the analogue
          text telephone protocol into ToIP and vice versa before the
          telephone switch connects the IP network.
          Note: The in band text dialogue is undesirable. This scenario
          SHOULD be avoided at any cost.
       
          Text gateway detection: prefix number or in band text signalling.
       
       9.10.11 VoIP user via a gateway/box connected to his/her own
       Broadband connection owning an analogue text telephone.
       
          The gateway box should natively transcode analogue text telephony
          into ToIP and vice versa when an analogue text phone is plugged in
          the RJ-11 socket [18].
       
          Text gateway detection: RJ-11 socket preconfigured by the box via
          jumpers or software.
       
       10. Terminal Features
       
          Implementers of products that support interactive Text-over-IP
          SHOULD NOT assume that all users of text are able to use
          mainstream input and output devices. People with arthritis or
          other dexterity problems might not be able to use very small
          keyboards. Visually impaired people might not be able to use
          standard sized characters on a display. Colour-blind people might
          suffer from badly chosen colour-schemes. People with motor
          disabilities might require specialised input devices.
       
          Implementers SHOULD try to make their products as open as possible
          with regard to this wide range of abilities and preferences and
          they MUST use standard interfaces wherever they provide such
          interfaces.
       
       
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       10.1 Text input
       
          Systems that support real-time interactive Text-over-IP SHOULD
          support suitable input mechanisms, either built-in or connectable
          through the use of a standard interface: PS/2, USB, Bluetooth, or
          virtual keyboard. In particular Braille users should be able to
          connect Braille keyboards to the terminal. Terminals MAY support a
          web interface for input and output of text.
       
          It is recommended that systems that fixed terminals that support
          real-time interactive Text-over-IP allow the user to enter the
          standard alphanumerical characters directly, rather than through a
          cycle of key presses or other indirect means. This could be done
          using full-sized keyboards, smaller sized keyboards or fastap
          keyboards for example. It is highly recommended to provide a
          standard interface to allow attachment of an external input
          device, especially for terminals that have only limited input
          systems built-in.
       
          All IP phones with a display of 12 or more characters MUST support
          at least text input through the regular phone keypad (and display
          of any incoming text) in order to provide basic emergency text
          communication from any IP phone.
       
          Input devices that have automatic key repeat MUST allow the user
          to specify the key-repeat rate.
       
       10.2 Text presentation
       
          Systems that support real-time interactive Text-over-IP SHOULD
          support suitable displays, either built-in or connectable through
          the use of a standard interface: S-VGA, USB, Bluetooth or IP.
          Braille readers should be connectable to the terminal using a
          standard interface.
       
          Terminals MAY support a web interface for input and output of
          text.
       
          While a variety of handsets and terminals might be developed for a
          number of equally varied scenarios, implementers MUST:
       
          In the case of fixed terminals or software applications on
          Personal Computers:
       
          a. Use either separate screen areas for displaying sent and
          received text OR clearly indicate the difference between sent and
          received text. Systems MAY allow the user to chose either on of
          these presentation methodologies.
       
          b. Provide at least 5 lines of 35 monospaced characters each for
          each direction (sent and received text) OR at least 10 lines of 35
          characters when sent and received text are presented together.
       
       
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          In the case of Mobile terminals:
       
          c. Use either separate screen areas for displaying sent and
          received text OR clearly indicate the difference between sent and
          received text. Systems MAY allow the user to chose either on of
          these presentation methodologies.
       
          d. Provide at least 3 lines of 20 monospaced characters each for
          each direction (sent and received text) OR at least 6 lines of 20
          characters when sent and received text are presented together.
       
          On both types of terminals, scrolling back through both sent and
          received text MUST be supported, including after the conversation
          has ended. Lines SHOULD be wrapped at word boundaries and this is
          strongly recommended.
       
          There MUST be an easy-to-use function to clear the screen at any
          time during the session, and if the implementation has chosen to
          present sent and received text separately, clearing the screen
          SHOULD be possible as a separate function for sent and received
          text.
       
          The function of the [CR], [LF] and [BACKSPACE] characters as
          explained in section 9.5. MUST be supported by the presentation.
          Presentation layers MUST support the full UTF-8 character set.
       
          When real-time Text-over-IP is used in conjunction with other
          modalities, like voice, the presentation MUST clearly indicate
          this to the user in an area outside the display region for send
          and received text.
       
          Identification information for other parties in the conversation,
          like URL’s, user-friendly names from an address book, or CLI in
          the case of conversations with text telephones, SHOULD be
          displayed throughout the entire conversation in a region outside
          the sent and received text area.
       
       10.3 Call control
       
          Call (Session) Control procedures MUST use the SIP protocol. Text
          sessions MUST be identified in accordance with requirements
          described earlier.
       
          Text services SHOULD be part of a Total Conversation environment
          in which voice, text and video sessions can be added, modified or
          deleted individually.
       
          To enable interworking with Textphones in telephone and cellular
          (mobile) networks, terminals MUST be able to access Gateways
          automatically when a PSTN or cellular (mobile) E.164-based
          telephone number is used as the called address.
       
       
       
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          Users MUST be able to establish text sessions to emergency service
          providers using the widely recognised emergency numbers in use in
          the country of operation of the terminal eg. ‘911’ in USA.
       
          The ability to transfer Location information SHALL be provided if
          the information is available from the terminal.
       
       10.4 Device control
       
          ToIP will support the text protocol stack described earlier and
          will require the use of RFC 2793 [5].  RFC 2793 defines the use of
          ITU-T T.140 [4] over RTP. T.140 is a text presentation protocol
          that is also used in the ITU-T H.series multimedia systems
          including some videoconferencing systems. It is also used by ITU-T
          V.18 [10], the Textphone interworking specification, and by the
          GSM and 3GPP text conversation specifications.
       
          ToIP will be a full-duplex service. Small displays may require the
          users to indicate (via text indications at the user level) that a
          user wishes to communicate in the half-duplex mode. This will
          require a signal to inform the other user to proceed eg. ‘GA’ as
          traditionally used by many half-duplex TTY users.
       
       10.5 Alerting
       
          The form of Alerting indication(s) provided to the user should be
          selectable to suit particular users. Alerting indications MAY
          include Sound, Tactile (eg. vibrational), Visual (on-screen
          symbols; separate flashing light), Motion (eg. movement of
          something).
       
          The ability to send an Alerting signal to an external interface
          SHOULD be provided. This will allow Alerting devices that are
          specific to users requirements to be attached.
       
          As many as possible of the following alternatives for alerting
          SHALL be provided:
              * Internal flash.
              * Two-pole connector for external alerting systems triggered
          by contact between the two poles when a ring signal is generated.
              * Bluetooth serial profile with AT command interface, sending
          the "RING" message, intended for a Bluetooth alerting receiver
          with flash, vibration or sound action.
              * SIP connected alerting device, that get its stimuli by being
          registered on the same sip address as the terminal.
       
       10.6 External interfaces
       
          Terminals for ToIP SHOULD provide external interfaces for the
          following functions:
              * Text input.
              * Text display.
              * Terminal control.
       
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              * Session control.
       
       10.7 Power
       
          As terminals could remain active for very long periods of time,
          the electrical power requirements of all the terminals SHOULD be
          as low as possible.
       
          If the terminal is to be used for calling Emergency services or
          where the mains power supply is unreliable, back-up power systems
          SHOULD be provided for the terminal and all equipment used to
          provide the ToIP service. This can be implemented in many
          different ways eg. via the line powering option on some Ethernet
          interfaces, or by using a ‘no break’ power supply (a battery back-
          up system with inverters that can recreate a limited amount of
          mains power).
       
       11. Security Considerations
       
          There are no additional security requirements other than described
          earlier.
       
       12. Authors’ Addresses
       
          The following people provided substantial technical and writing
          contributions to this document, listed alphabetically:
       
          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
          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
          Sweden
          Phone: +46 708 204 288 / +46 8 556 002 03
          Fax:   +46 8 556 002 06
          Email: gunnar.hellstrom@omnitor.se
       
          Paul E. Jones
       
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          Cisco Systems, Inc.
          7025 Kit Creek Rd.
          Research Triangle Park, NC 27709
          Phone: +1 919 392 6948
          E-mail: paulej@packetizer.com
       
          Radhika R. Roy
          AT&T
          Room C1-2B03
          200 Laurel Avenue S.
          Middletown, NJ 07748
          USA
          Phone: +1 732 420 1580
          Fax: +1 732 368 1302
          Email: rrroy@att.com
       
          Henry Sinnreich
          MCI
          400 International Parkway
          Richardson, Texas 75081
          Email: henry.sinnreich@mci.com
       
          Gregg C Vanderheiden
          University of Wisconsin-Madison
          Trace R & D Center
          1550 Engineering Dr (Rm 2107)
          Madison, Wi  53706
          USA
          gv@trace.wisc.edu
          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
       
       13. Full Copyright Statement
       
          Copyright (C) The Internet Society (1999, 2000). All Rights
          Reserved. This document and translations of it may be copied and
          furnished to others, and derivative works that comment on or
          otherwise explain it or assist in its implementation may be
          prepared, copied, published and distributed, in whole or in part,
          without restriction of any kind, provided that the above copyright
          notice and this paragraph are included on all such copies and
          derivative works. However, this document itself may not be
          modified in any way, such as by removing the copyright notice or
          references to the Internet Society or other Internet
       
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          organizations, except as needed for the purpose of developing
          Internet standards in which case the procedures for copyrights
          defined in the Internet Standards process must be followed, or as
          required to translate it into languages other than English.
       
          The limited permissions granted above are perpetual and will not
          be revoked by the Internet Society or its successors or assigns .
          This document and the information contained herein is provided on
          an "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET
          ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR
          IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
          THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
          WARRANTIES OF MERCHANTABILITY OR FIT-NESS FOR A PARTICULAR
          PURPOSE."
       
       14. References
       
       14.1 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. Hellström, ”RTP Payload for Text Conversation, RFC 2793, May
          2000.
          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 august 2003 -  Work in progress.
          8. G. Camarillo, H. Schulzrinne, E. Burger, and A. Wijk,
          “Transcoding Service Invocation in SIP using Third Party Call
          Control,” IETF, August 2003 - Work in Progress.
          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
          2000.
          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.
       
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          15. TIA/EIA/IS-823-A  “TTY/TDD Extension to TIA/EIA-136-410
          Enhanced Full Rate Speech Codec (must used in conjunction with
          TIA/EIA/IS-840)”
          16. TIA/EIA/IS-127-2  ‘Enhanced Variable Rate Codec, Speech
          Service Option 3 for Wideband Spread Spectrum Digital Systems.
          Addendum 2.”
          17. 3GPP TS26.226  “Cellular Text Telephone Modem Description”
          (CTM).
          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.
       
       
       14.2 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
          V5.0.0, March 2001"SIP Telephony Device Requirements,
          Configuration and Data" by manyfolks,IETF, October 2003.
       
       
       
       
       
       
       
       
       
       
       
       
       
       
       
       
       
       
       
       
       
       
       
       
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