[Docs] [txt|pdf] [Tracker] [WG] [Email] [Diff1] [Diff2] [Nits]

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

   SIPPING Workgroup
   Internet Draft                                           A. van Wijk
   Category: Informational                                       AnnieS
   Expires: September 5 2006                              March 6, 2006


             Framework for real-time text over IP using SIP

                     draft-ietf-sipping-toip-04.txt

Status of this Memo

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

   Internet-Drafts are working documents of the Internet Engineering
   Task 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.

   This Internet-Draft will expire on September 5, 2006.

Copyright Notice

   Copyright (C) The Internet Society (2006).

Abstract

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



A. van Wijk, et al.    Expires September 5 2006               [Page 1]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


Table of Contents

   1. Introduction...................................................3
   2. Scope..........................................................4
   3. Terminology....................................................4
   4. Definitions....................................................4
   5. Requirements...................................................6
      5.1 General requirements for ToIP..............................6
      5.2 Detailed requirements for ToIP.............................8
         5.2.1 Session control and set-up requirements...............8
         5.2.2 Transport requirements................................9
         5.2.3 Transcoding service requirements.....................10
         5.2.4 Presentation and User control requirements...........11
         5.2.5 Interworking requirements............................12
            5.2.5.1 PSTN Interworking requirements..................12
            5.2.5.2 Cellular Interworking requirements..............12
            5.2.5.3 Instant Messaging Interworking requirements.....13
   6. Implementation Framework......................................13
      6.1 Framework of general implementation.......................13
      6.2 Framework of detailed implementation......................14
         6.2.1 Session control and set-up...........................14
            6.2.1.1 Pre-session setup...............................14
            6.2.1.2 Basic Point-to-Point Session setup..............15
            6.2.1.3 Addressing......................................15
            6.2.1.4 Session Negotiations............................15
            6.2.1.5 Additional session control......................16
         6.2.2 Transport............................................16
         6.2.3 Transcoding services.................................17
         6.2.4 Presentation and User control functions..............18
            6.2.4.1 Progress and status information.................18
            6.2.4.2 Alerting........................................18
            6.2.4.3 Answering Machine...............................18
            6.2.4.4 Text presentation...............................19
            6.2.4.5 File storage....................................19
         6.2.5 Interworking functions...............................19
            6.2.5.1 PSTN Interworking...............................20
            6.2.5.2 Mobile Interworking.............................21
               6.2.5.2.1 Cellular "No-gain".........................21
               6.2.5.2.2 Cellular Text Telephone Modem (CTM)........21
               6.2.5.2.3 Cellular "Baudot mode".....................22
               6.2.5.2.4 Mobile data channel mode...................22
               6.2.5.2.5 Mobile ToIP................................22
            6.2.5.3 Instant Messaging Interworking..................22
            6.2.5.4 Interworking through gateways...................23
            6.2.5.5 Multi-functional Combination gateways...........24
            6.2.5.6 Character set transcoding.......................25
   7. Further recommendations for implementers and service providers25
      7.1 Access to Emergency services..............................25
      7.2 Home Gateways or Analog Terminal Adapters.................26


A. van Wijk, et al.    Expires September 5 2006               [Page 2]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


      7.3 User Mobility.............................................26
      7.4 Firewalls and NATs........................................26
   8. IANA Considerations...........................................26
   9. Security Considerations.......................................26
   10. Authors’ Addresses...........................................27
   11. References...................................................28
      11.1 Normative references.....................................28
      11.2 Informative references...................................30


1.
  Introduction

   For many years, text has been in use as a medium for conversational,
   interactive dialogue between users in a similar way to how voice
   telephony is used. Such interactive text is different from messaging
   and semi-interactive solutions like Instant Messaging in that it
   offers an equivalent conversational experience to users who cannot,
   or do not wish to, use voice. It therefore meets a different set of
   requirements from other text-based solutions already available on IP
   networks.

   Traditionally, deaf, hard of hearing and speech-impaired people are
   amongst the most prolific users of conversational, interactive text
   but, because of its interactivity, it is becoming popular amongst
   mainstream users as well.

   This document describes how existing IETF protocols can be used to
   implement a Text-over-IP solution (ToIP). This ToIP framework is
   specifically designed to be compatible with Voice-over-IP (VoIP) and
   Multimedia-over-IP (MoIP) environments, as well as meeting the user’s
   requirements, including those of deaf, hard of hearing and speech-
   impaired users as described in RFC3351 [2] and mainstream users.

   The Session Initiation Protocol (SIP) [3] is the protocol of choice
   for control of Multimedia communications and Voice-over-IP (VoIP) in
   particular. It offers all the necessary control and signaling
   required for the ToIP framework.

   The Real-Time Transport Protocol (RTP) [4] is the protocol of choice
   for real-time data transmission, and its use for real-time text
   payloads is described in RFC4103 [5].

   This document defines a framework for ToIP to be used either by
   itself or as part of integrated, multi-media services, including
   Total Conversation [6].






A. van Wijk, et al.    Expires September 5 2006               [Page 3]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


2.
  Scope

   This document defines a framework for the implementation of real-time
   ToIP, either stand-alone or as a part of multimedia services,
   including Total Conversation [6]. It defines the:

     a. Requirements of Real-time text;
     b. Requirements for ToIP interworking;
     c. Description of ToIP implementation using SIP and RTP;
     d. Description of ToIP interworking with other text services.

3.
  Terminology

   In this document, the key words "MUST", "MUST NOT", "REQUIRED",
   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT
   RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as
   described in BCP 14, RFC 2119 [7] and indicate requirement levels for
   compliant implementations.

4.
  Definitions

   Audio bridging: a function of an audio media bridge server, gateway
   or relay service that bridges audio into a single source through
   combining audio from multiple users excluding each destination
   source’s audio and sends to each respective destination enabling an
   audio path through the service between the users involved in the
   call.

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

   Full duplex: media is sent independently in both directions.

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

   Interactive text: a term for real time transmission of text in a
   character-by-character fashion for use in conversational services,
   often as a text equivalent to voice based conversational services.
   (Equivalent to real-time text.)

   Real-time text: a term for real time transmission of text in a
   character-by-character fashion for use in conversational services,
   often as a text equivalent to voice based conversational services.
   Conversational text is defined in ITU-T F.700 Framework for
   multimedia services [25].



A. van Wijk, et al.    Expires September 5 2006               [Page 4]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


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

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

   Text bridging: a function of a gateway service that enables the flow
   of text through the service between the users involved in the call.

   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
   real-time text in a call.

   Text Bridging: a function of the text media bridge server, gateway or
   relay service that bridges real-time text into a single source
   through combining real-time text from multiple users excluding each
   destination source’s real-time text and sends to each respective
   destination enabling a real-time text path through the service
   between the users involved in the call.

   Text telephony: analog textphone service.

   Total Conversation: a multimedia service offering real time
   conversation in video, real-time text and voice according to
   interoperable standards. All media flow in real time. (See ITU-T
   F.703 "Multimedia conversational services" [6].)

   Transcoding Services: services of a third-party user agent that
   transcodes one stream into another. Transcoding can be done by human
   operators, in an automated manner or a combination of both methods.
   Text Relay Services are examples of a transcoding service between
   real-time text and audio.

   TTY: alternative designation for a text telephone or textphone, often
   used in USA. Also called TDD, Telecommunication Device for the Deaf.

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





A. van Wijk, et al.    Expires September 5 2006               [Page 5]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   Acronyms:

   2G     Second generation cellular (mobile)
   2.5G   Enhanced second generation cellular (mobile)
   3G     Third generation cellular (mobile)
   CDMA   Code Division Multiple Access
   CLI    Calling Line Identification
   CTM    Cellular Text Telephone Modem
   ENUM   E.164 number storage in DNS (see RFC3761)
   GSM    Global System of Mobile Communication
   ISDN   Integrated Services Digital Network
   ITU-T  International Telecommunications Union-Telecommunications
          Standardisation Sector
   NAT    Network Address Translation
   PSTN   Public Switched Telephone Network
   RTP    Real Time Transport Protocol
   SDP    Session Description Protocol
   SIP    Session Initiation Protocol
   SRTP   Secure Real Time Transport Protocol
   TDD    Telecommunication Device for the Deaf
   TDMA   Time Division Multiple Access
   TTY    Analog textphone (Teletypewriter)
   ToIP   Real-time Text over Internet Protocol
   UTF-8  Universal Transfer Format-8
   VCO/HCO Voice Carry Over/Hearing Carry Over
   VoIP   Voice over Internet Protocol

5.
  Requirements

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

   Real-time text conversation can be combined with other conversational
   services like video or voice.

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

   This section (Requirements) informs implementers about WHICH
   requirements the systems and services shall meet. The next section
   (Section 6 Framework Implementation) describes HOW to do it.

5.1
   General requirements for ToIP

   Any framework for ToIP must be designed to meet the requirements of
   RFC3351 [2]. A basic requirement is that it must provide a


A. van Wijk, et al.    Expires September 5 2006               [Page 6]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   standardized way for offering text-based, conversational services
   that can be used as an equivalent to voice telephony by deaf, hard of
   hearing speech-impaired and mainstream users.

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

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

   In order to make ToIP the text equivalent of voice services, it needs
   to offer equivalent features in terms of conversationality as voice
   telephony provides. To achieve that, ToIP needs to:

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

   Real-time text is a useful subset of Total Conversation defined in
   ITU-T F.703 [6]. Users could use multiple modes of communication
   during the conversation, either at the same time or by switching
   between modes, e.g., between real-time text and audio.

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

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


A. van Wijk, et al.    Expires September 5 2006               [Page 7]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006



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

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

5.2
   Detailed requirements for ToIP

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

5.2.1
     Session control and set-up requirements

   Users will set up a session by identifying the remote party or the
   service they want to connect to. However, conversations could be
   started using a mode other than the real-time text.

   Simultaneous or alternating use of voice and real-time text is used
   by a large number of users who can send voice but must receive text
   (due to a hearing impairment), or who can hear but must send text
   (due to a speech impairment).

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

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

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

   R4: 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.

   R5: If the user requests simultaneous use of real-time text and
   audio, and this is not possible either because the system only
   supports alternate modalities or because of constraints in the



A. van Wijk, et al.    Expires September 5 2006               [Page 8]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   network, the system MUST try to establish communication with best
   effort.

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

   R7: It SHOULD be possible to use the real-time text medium in
   conference sessions in a similar way to how audio is handled and
   video is displayed.

   Real-time text in conferences can be used both for letting individual
   participants use the text medium (for example, for sidebar
   discussions in text while listening to the main conference audio), as
   well as for central support of the conference with real time text
   interpretation of speech.

   R8: During session set up, it SHOULD be possible for the users to
   indicate if the caller wants to use voice and real-time text
   simutaneously as part of the conversation.

   R9: Session set up and negotiation of modalities must allow users to
   specify the language of the real-time text to be used. (It is
   recommended that similar functionality is provided for the video part
   of the conversation, i.e. to specify the sign language being used).

5.2.2
     Transport requirements

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

   R10: Characters must be transmitted soon after entry of each
   character so that the maximum delay requirement can be met. A delay
   time of one second is regarded good, while a delay of two seconds is
   possible to use.

   R11: It must be possible to transmit characters at a rate sufficient
   to support fast human typing as well as speech to text methods of
   generating conversation text. A rate of 20 characters per second is
   regarded sufficient.

   R12: a ToIP service must be able to deal with international character
   sets.




A. van Wijk, et al.    Expires September 5 2006               [Page 9]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   R13: Where it is possible, loss of real-time text during transport
   should be detected and the user should be informed.

   R14: Transport of real-time text should be as robust as possible, so
   as to minimize loss of characters.

   R15: Where possible, it must be possible to send and receive real-
   time text simultaneously.

5.2.3
     Transcoding service requirements

   If the User Agents of different participants indicate that there is
   an incompatibility between their capabilities to support certain
   media types, e.g. one terminal only offering T.140 over IP as
   described in RFC4103 [5] and the other one only supporting audio, the
   user might want to invoke a transcoding service.

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

   A number of requirements, motivations and implementation guidelines
   for relay service invocation can be found in RFC 3351 [2].

   R16: It MUST be possible for users to invoke a transcoding service
   where such service is available.

   R17: It MUST be possible for users to indicate their preferred
   modality.

   R18: The requirements for transcoding services need to be negotiated
   in real-time to set up the session.

   R19: Adding or removing a relay service MUST be possible without
   disrupting the current session.

   R20: When setting up a session, it MUST be possible for a user to
   determine the type of relay service requested (e.g., speech to text
   or text to speech). The specification of a type of relay MUST include
   a language specifier.

   R21: It SHOULD be possible to route the session to a preferred relay
   service even if the user invokes the session from another region or
   network than that usually used.




A. van Wijk, et al.    Expires September 5 2006              [Page 10]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


5.2.4
     Presentation and User control requirements

   R22: User Agents for ToIP services must have alerting methods (e.g.,
   for incoming sessions) that can be used by deaf and hard of hearing
   people or provide a range of alternative, but equivalent, alerting
   methods that can be selected by all users, regardless of their
   abilities.

   R23: Where real-time text is used in conjunction with other media,
   exposure of user control functions through the User Interface needs
   to be done in an equivalent manner for all supported media.

   In other words, where certain call control functions are available
   for the audio media part of a session, these functions MUST also be
   supported for the real-time text media part of the same session. For
   example, call transfer must act on all media in the session.

   R24: If present, identification of the originating party (for example
   in the form of a URL or a CLI) MUST be clearly presented to the user
   in a form suitable for the user BEFORE the session invitation is
   answered.

   R25: When a session invitation involving ToIP originates from a PSTN
   text telephone (e.g. transcoded via a text gateway), this SHOULD be
   indicated to the user. The ToIP client MAY adjust the presentation of
   the real-time text to the user as a consequence.

   R26: An indication should be given to the user when real-time text is
   available during the call, even if it is not invoked at call setup
   (e.g. when only voice and/or video is used initially).

   R27: The user MUST be informed of any change in modalities.

   R28: Users must be presented with appropriate session progress
   information at all times.

   R29: Answering machine functions SHOULD be provided by the User
   Agent.

   R30: When the answering machine function is enabled on the User
   Agent, alerting of the user SHOULD still be possible and users SHOULD
   be able to take over control from the answering machine function at
   any time.

   R31: Users SHOULD be able to save the text portion of a conversation.

   R32: The presentation of the conversation should be done in such a
   way that users can easily identify which party generated any given
   portion of text.


A. van Wijk, et al.    Expires September 5 2006              [Page 11]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


5.2.5
     Interworking requirements

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

   Real-time/Interactive texting facilities exist already in various
   forms and on various networks. On the PSTN, it is commonly referred
   to as text telephony.

   Text gateways are used for converting between different media types.
   They could be used between networks or within networks where
   different transport technologies are used.

   R33: ToIP SHOULD provide interoperability with text conversation
   features in other networks, for instance the PSTN.

   R34: When communicating via a gateway to other networks and
   protocols, the ToIP service SHOULD support the functionality for
   alternating or simultaneous use of modalities as offered by the
   interworking network.

   R35: Address information, both called and calling, SHOULD be
   transferred, and possibly converted, when interworking between
   different networks.

   R36: When interworking with other networks and services, the ToIP
   service SHOULD provide buffering mechanisms to deal with delays in
   call setup, transmission speeds and/or to interwork with half duplex
   services.

5.2.5.1
       PSTN Interworking requirements

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

   R37: ToIP services MUST provide interworking with PSTN legacy text
   telephony devices.

   R38: When interworking with PSTN legacy text telephony services,
   alternating text and voice function MAY be supported. (Called "voice
   carry over (VCO) and hearing carry over (HCO)").

5.2.5.2
       Cellular Interworking requirements

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



A. van Wijk, et al.    Expires September 5 2006              [Page 12]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   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 [8] 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 ToIP at the
   other side.

   R39: a ToIP service SHOULD provide interworking with mobile text
   conversation services.

5.2.5.3
       Instant Messaging Interworking requirements

   Many people use Instant Messaging to communicate via the Internet
   using text. Instant Messaging usually transfers blocks of text rather
   than streaming as is used by ToIP. Usually a specific action is
   required by the user to activate transmission, such as pressing the
   ENTER key or a send button. As such, it is not a replacement for ToIP
   and in particular does not meet the needs for real time conversations
   including those of deaf, hard of hearing and speech-impaired users as
   defined in RFC 3351 [2]. It is unsuitable for communications through
   a relay service [I]. The streaming nature of ToIP provides a more
   direct conversational user experience and, when given the choice,
   users may prefer ToIP.

   R39: a ToIP service MAY provide interworking with Instant Messaging
   services.

6.
  Implementation Framework

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

6.1
   Framework of general implementation

   ToIP uses the Session Initiation Protocol (SIP) [3] to set up,
   control and tear down the connections between users whilst the media
   is transported using the Real-Time Transport Protocol (RTP) [4] as
   described in RFC4103 [5].

   SIP [3] allows participants to negotiate all media including real-
   time text conversation [5]. This is a highly desirable function for


A. van Wijk, et al.    Expires September 5 2006              [Page 13]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   all IP telephony users but essential for deaf, hard of hearing, or
   speech impaired people who have limited or no use of the audio path
   of the call. Even for mainstream users, media negotiations like real-
   time text are also very useful in many circumstances as described
   earlier.

   The ability of SIP to set up conversation sessions from any location,
   as well as its privacy and security provisions, MUST be maintained by
   ToIP services.

   Real-time text conversation based on the presentation protocol T.140
   [9], in addition to audio and video communications, is a valuable
   service for many users, including those on non-IP networks. T.140
   also provides for basic real-time editing of the text.

6.2
   Framework of detailed implementation

6.2.1
     Session control and set-up

   ToIP services MUST use the Session Initiation Protocol (SIP) [3] for
   setting up, controlling and terminating sessions for real-time text
   conversation with one or more participants and possibly including
   other media like video or audio. The session description protocol
   (SDP) used in SIP to describe the session is used to express the
   attributes of the session and to negotiate a set of compatible media
   types.

6.2.1.1
       Pre-session setup

   The requirements of the user to be reached at a consistent address
   and to store preferences for evaluation at session setup are met by
   pre-session setup actions. That includes storing of registration
   information in the SIP registrar, to provide information about how a
   user can be contacted. This will allow sessions to be set up rapidly
   and with proper routing and addressing.

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

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

   b. Server support of User Preferences: SIP servers that support ToIP
   services MUST have the capability to act on calling user preferences
   for real-time text in order to accept or reject the session.The
   actions taken can be based on the called user’s preferences defined
   as part of the pre-session setup registration. For example, if the


A. van Wijk, et al.    Expires September 5 2006              [Page 14]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   user is called by another party, and it is determined that a
   transcoding server is needed, the session should be re-directed or
   otherwise handled accordingly.

6.2.1.2
       Basic Point-to-Point Session setup

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

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

   a. Caller Preferences: SIP headers (e.g., Contact)[11] can be used to
   show that ToIP is the medium of choice for communications.

   b. Called Party Preferences [12]: The called party being passive can
   formulate a clear rule indicating how a session should be handled
   either using real-time text as a preferred medium or not, and whether
   a designated SIP proxy needs to handle this session or it will be
   handled in the SIP user agent.

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

6.2.1.3
       Addressing

   The SIP [3] addressing schemes MUST be used for all entities in a
   ToIP session. For example, SIP URL’s or Tel URL’s are used for
   caller, called party, user devices, and servers (e.g., SIP server,
   Transcoding server).

6.2.1.4
       Session Negotiations

   The Session Description Protocol (SDP) used in SIP [3] provides the
   capabilities to indicate real-time text as a medium in the session
   setup. RFC 4103 [5] uses the RTP payload types "text/red" and
   "text/t140" for support of ToIP which can be indicated in the SDP as
   a part of the SIP INVITE, OK and SIP/200/ACK media negotiations. In
   addition, SIP’s offer/answer model [13] can also be used in
   conjunction with other capabilities including the use of a
   transcoding server for enhanced session negotiations [14,15,16].

   Systems SHOULD provide a best-effort approach to answering
   invitations for session set-up and users SHOULD be informed when the


A. van Wijk, et al.    Expires September 5 2006              [Page 15]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   session is accepted by the other party. On all systems that both
   inform users of session status and support ToIP, this information
   MUST be available in textual form and MAY also be provided in other
   media.

6.2.1.5
       Additional session control

   Systems that support additional session control features, for example
   call waiting, forwarding, hold etc on voice sessions, MUST offer this
   functionality for text sessions.

6.2.2
     Transport

  A ToIP service MUST always support at least one real-time text media
  type.

   ToIP services MUST support the Real-Time Transport Protocol (RTP) [4]
   according to the specification of RFC4103 [4] for the transport of
   text between participants.

   RFC4103 describes the transmission of T.140 [9] real-time text on IP
   networks.

   In order to enable the use of international character sets, the
   transmission format for text conversation SHALL be UTF-8 [17], in
   accordance with ITU-T T.140.

   If real-time text is detected to be missing after transmission, there
   SHOULD be a "text loss" indication in the real-time text as specified
   in T.140 Addendum 1 [9].

   ToIP uses RTP as the default transport protocol for the transmission
   of real-time text via the medium "text/t140" as specified in RFC 4103
   [5].

   The redundancy method of RFC 4103 [5] SHOULD be used to significantly
   increase the reliability of the real-time text transmission. A
   redundancy level using 2 generations gives very reliable results and
   is therefore strongly RECOMMENDED.

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

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




A. van Wijk, et al.    Expires September 5 2006              [Page 16]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   By having this single coding and transmission scheme for real time
   text defined in the SIP session control environment, the opportunity
   for interoperability is optimized. However, if good reasons exist,
   other transport mechanisms MAY be offered and used for the T.140
   coded text provided that proper negotiation is introduced, but RFC
   4103 [5] transport MUST be used as both the default and the fallback
   transport.

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

   The text transmission SHALL allow a rate of at least 30 characters
   per second.

6.2.3
     Transcoding services

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

   A specific type of transcoding service in a ToIP environment is a
   relay service. The relay service acts as an intermediary between two
   or more callers using different media or different media encoding
   schemes.

   The basic text relay service allows a translation of speech to real-
   time text and real-time text to speech, which enables hearing and
   speech impaired callers to communicate with hearing callers. Even
   though this document focuses on ToIP, we want to remind readers that
   other relay services exist, like video relay services transcoding
   speech to sign language and vice versa where the signing is
   communicated using video.

   It is RECOMMENDED that ToIP implementations make the invocation and
   use of relay services as easy as possible. It MAY happen
   automatically when the session is being set up based on any valid
   indication or negotiation of supported or preferred media types. A
   transcoding framework document using SIP [14] describes invoking
   relay services, where the relay acts as a conference bridge or uses
   the third party control mechanism. ToIP implementations SHOULD
   support this transcoding framework.








A. van Wijk, et al.    Expires September 5 2006              [Page 17]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


6.2.4
     Presentation and User control functions

6.2.4.1
       Progress and status information

   During a conversation that includes ToIP, status and session progress
   information MUST be provided in a textual form so users can perform
   all session control functions. That information MUST be equivalent to
   session progress information delivered in any other format, for
   example audio.

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

   There MUST be a clear indication, in a modality useful to the user,
   whenever a session is connected or disconnected. A user SHOULD never
   be in doubt about the status of the session, even if the user is
   unable to make use of the audio or visual indication. For example,
   tactile indications could be used by deafblind individuals.

   In summary, it SHOULD be possible to observe indicators about:

   - Incoming session
   - Availability of real-time text, voice and video channels
   - Session progress
   - Incoming real-time text
   - Any loss in incoming real-time text
   - Typed and transmitted real-time text.

6.2.4.2
       Alerting

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

   Among the alerting options are alerting by the User Agent’s User
   Interface and specific alerting user agents registered to the same
   registrar as the main user agent.

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

6.2.4.3
       Answering Machine

   Systems for ToIP MAY support an answering machine function,
   equivalent to answering machines on telephony networks. If an
   answering machine function is supported, it MUST support at least 160


A. van Wijk, et al.    Expires September 5 2006              [Page 18]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   characters for the greeting message. It MUST support incoming real-
   time text message storage of a minimum of 4096 characters, although
   systems MAY support much larger storage. It is RECOMMENDED that
   systems support storage of at least 20 incoming messages of up to
   16000 characters per message.

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

6.2.4.4
       Text presentation

   When the display of text conversation is included in the design of
   the end user equipment, the display of the dialogue SHOULD be made so
   that it is easy to differentiate the text belonging to each party in
   the conversation. This could be done using color, positioning of the
   text (i.e. incoming real-time text and outgoing real-time text in
   different display areas), by in-band identifiers of the parties or by
   a combination of any of these techniques.

   ToIP SHOULD handle characters such as new line, erasure and alerting
   during a session as specified in ITU-T T.140 [9].

6.2.4.5
       File storage

   Systems that support ToIP MAY save the text conversation to a file.
   This SHOULD be done using a standard file format. For example: a UTF8
   text file in XHTML format [18] including timestamps, party names (or
   addresses) and the text conversation.

6.2.5
     Interworking functions

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

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




A. van Wijk, et al.    Expires September 5 2006              [Page 19]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


6.2.5.1
       PSTN Interworking

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

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

   In particular transmission of interactive text on PSTN networks takes
   place using a variety of codings and modulations, including ITU-T
   V.21 [III], Baudot [II], DTMF, V.23 [IV] and others. Many
   difficulties have arisen as a result of this variety in text
   telephony protocols and the ITU-T V.18 [19] standard was developed to
   address some of these issues.

   ITU-T V.18 [19] offers a native text telephony method plus it defines
   interworking with current protocols. In the interworking mode, it
   will recognise one of the older protocols and fall back to that
   transmission method when required.

   Text gateways MUST use the ITU-T V.18 [19] standard at the PSTN side.
   A text gateway MUST act as a SIP User Agent on the IP side and
   support RFC4103 text transport.

   PSTN-ToIP gateways MUST allow alternating use of real-time text and
   voice if the PSTN textphone involved at the PSTN side of the session
   supports this. (This mode is often called VCO/HCO).

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

   While ToIP allows receiving and sending real-time text simultaneously
   and is displayed on a split screen, many analog text telephones
   require users to take turns typing.
   This is because many text telephones operate strictly half duplex.
   Only one can transmit text at a time. The users apply strict turn-
   taking rules.



A. van Wijk, et al.    Expires September 5 2006              [Page 20]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   There are several text telephones which communicate in full duplex,
   but merge transmitted text and received text in the same line in the
   same display window. And also here do the users apply strict turn
   taking rules.
   Native V.18 text telephones support full duplex and separate display
   from reception and transmission so that the full duplex capability
   can be used fully. Such devices could use the ToIP split screen as
   well, but almost all text telephones use a restricted character set
   and many use low text transmission speeds (4 to 7 charcters per
   second).

   That is why it is important for the ToIP user to know that he or she
   is connected with an analog text telephone. The "txp" media content
   attribute [10]SHOULD be used to indicate that the call originates
   from a PSTN text telephone (e.g. via an ATA or a text gateway).

6.2.5.2
       Mobile Interworking

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

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

   The following sections describe how mobile text telephony is
   supported.

6.2.5.2.1
         Cellular "No-gain"

   The "No-gain" text telephone transporting technology uses specially
   modified EFR [20] and EVR [21] speech vocoders in mobile terminals
   used to provide a text telephony call. It provides full duplex
   operation and supports alternating voice and text ("VCO/HCO"). It is
   dedicated to CDMA and TDMA mobile technologies and the US Baudot
   (i.e. 45 bit/s) type of text telephones.

6.2.5.2.2
         Cellular Text Telephone Modem (CTM)

   CTM [8] is a technology independent modem technology that provides
   the transport of text telephone characters at up to 10 characters/sec
   using modem signals that can be carried by many voice codecs and uses
   a highly redundant encoding technique to overcome the fading and cell
   changing losses.






A. van Wijk, et al.    Expires September 5 2006              [Page 21]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


6.2.5.2.3
         Cellular "Baudot mode"

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

6.2.5.2.4
         Mobile data channel mode

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

6.2.5.2.5
         Mobile ToIP

   ToIP could be supported over mobile wireless packet switched services
   that interface to the Internet. For 3GPP 3G services, ToIP support is
   described in 3G TS 26.235 [22].

6.2.5.3
       Instant Messaging Interworking

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

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

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

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

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


A. van Wijk, et al.    Expires September 5 2006              [Page 22]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


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

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

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

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

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

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

6.2.5.4
       Interworking through gateways

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

   Text gateways MUST allow for the differences that result from
   different text protocols. The protocols to be supported will depend
   on the service requirements of the Gateway.

   Session setup through gateways to other networks MAY require the use
   of specially formatted addresses or other mechanisms for invoking
   those gateways.

   Different data rates of different protocols MAY require text
   buffering.


A. van Wijk, et al.    Expires September 5 2006              [Page 23]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006



   When text gateway functions are invoked, there will be a need for
   intermediate storage of characters before transmission to a device
   receiving text slower than the transmitting speed of the sender. Such
   temporary storage SHALL be dimensioned to adjust for receiving at 30
   characters per second and transmitting at 6 characters per second for
   up to 4 minutes (i.e. less than 3000 characters).

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

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

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

6.2.5.5
       Multi-functional Combination gateways

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

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

   - PSTN text telephony to ToIP.
   - PSTN text telephony to Instant Messaging.


A. van Wijk, et al.    Expires September 5 2006              [Page 24]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   - Instant Messaging to ToIP.

6.2.5.6
       Character set transcoding

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

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

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

7.
  Further recommendations for implementers and service providers

7.1
   Access to Emergency services

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

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

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







A. van Wijk, et al.    Expires September 5 2006              [Page 25]
Internet-Draft    Framework for real-time ToIP using SIP    March 2006


7.2
   Home Gateways or Analog Terminal Adapters

   Analog terminal adapters (ATA) using SIP based IP communication and
   RJ-11 connectors for connecting traditional PSTN devices SHOULD
   enable connection of legacy PSTN text telephones [23].

   These adapters SHOULD contain V.18 modem functionality, voice
   handling functionality, and conversion functions to/from SIP based
   ToIP with T.140 transported according to RFC 4103 [4], in a similar
   way as it provides interoperability for voice sessions.

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

   NOTE: It is preferred that such analog terminal adaptors do use RFC
   4103 [5] on board and thus act as a text gateway. Sending textphone
   signals over the voice channel is undesirable due to possible
   filtering and compression and packet loss between the end-points.
   This can result in character loss in the textphone conversation or
   even not allowing the textphones to connect to each other.

7.3
   User Mobility

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

7.4
   Firewalls and NATs

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

8.
  IANA Considerations

   There are no IANA considerations for this specification.

9.
  Security Considerations

   User confidentiality and privacy need to be met as described in SIP
   [3]. For example, nothing should reveal the fact that the ToIP user
   might be a person with a hearing or speech impairment. ToIP is after
   all a mainstream communication medium for all users. It is up to the



A. van Wijk, et al.    Expires September 5 2006              [Page 26]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   ToIP user to make his or her hearing or speech impairment public. If
   a transcoding server is being used, this SHOULD be transparent.
   Encryption SHOULD be used on end-to-end or hop-by-hop basis as
   described in SIP [3] and SRTP [24].

   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.

10.
   Authors’ Addresses

   The following people provided substantial technical and writing
   contributions to this document, listed alphabetically:

   Willem Dijkstra
   TNO Informatie- en Communicatietechnologie
   Eemsgolaan 3
   9727 DW Groningen
   tel  : +31 50 585 77 24
   fax  : +31 50 585 77 57
   Email: willem.dijkstra@tno.nl

   Barry Dingle
   ACIF, 32 Walker Street
   North Sydney, NSW 2060 Australia
   Tel +61 (0)2 9959 9111
   Mob +61 (0)41 911 7578
   Email: btdingle@gmail.com

   Guido Gybels
   Department of New Technologies
   RNID, 19-23 Featherstone Street
   London EC1Y 8SL, UK
   Tel +44(0)20 7294 3713
   Txt +44(0)20 7608 0511
   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


A. van Wijk, et al.    Expires September 5 2006              [Page 27]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006



   Radhika R. Roy
   SAIC
   3465-B Box Hill Corporate Center Drive
   Abingdon, MD 21009
   Tel: 443 402 9041
   Email: Radhika.R.Roy@saic.com

   Henry Sinnreich
   pulver.com
   115 Broadhollow Rd
   Suite 225
   Melville, NY 11747
   USA
   Tel: +1.631.961.8950

   Gregg C Vanderheiden
   University of Wisconsin-Madison
   Trace R & D Center
   1550 Engineering Dr (Rm 2107)
   Madison, Wi  53706
   USA
   Phone +1 608 262-6966
   FAX +1 608 262-8848
   Email: gv@trace.wisc.edu

   Arnoud A. T. van Wijk
   Foundation for an Information and Communication Network for the Deaf
   and Hard of Hearing
   "AnnieS"
   www.annies.nl
   Email: arnoud@annies.nl

11.
   References

11.1
    Normative references

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

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

   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.



A. van Wijk, et al.    Expires September 5 2006              [Page 28]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


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

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

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

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

   8.  3GPP TS 26.226  "Cellular Text Telephone Modem Description"
       (CTM).

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

   10. J. Hautakorpi, G. Camarillo, "The SDP (Session Description
       Protocol) Content Attribute", IETF, February 2006 - Work in
       Progress.

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

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

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

   14. G. Camarillo, "Framework for Transcoding with the Session
       Initiation Protocol" IETF Nov 2005 -  Work in progress.

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

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

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



A. van Wijk, et al.    Expires September 5 2006              [Page 29]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


   18. "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.

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

   20. 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)"

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

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

   23. H. Sinnreich, S. Lass,  and C. Stredicke, "SIP Telephony Device
       Requirements and Configuration," IETF, October 2005 - Work in
       Progress.

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

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

11.2
    Informative references

   I. A relay service allows the users to transcode between different
   modalities or languages. In the context of this document, relay
   services will often refer to text relays that transcode text into
   voice and vice-versa. See for example http://www.typetalk.org.

   II. 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.)

   III. 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.

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





A. van Wijk, et al.    Expires September 5 2006              [Page 30]

Internet-Draft    Framework for real-time ToIP using SIP    March 2006


Full Copyright Statement

   Copyright (C) The Internet Society (2006).

   This document is subject to the rights, licenses and restrictions
   contained in BCP 78, and except as set forth therein, the authors
   retain all their rights.

   This document and the information contained herein are provided on
   an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE
   REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE
   INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR
   IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
   THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
   WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Intellectual Property

   The IETF takes no position regarding the validity or scope of any
   Intellectual Property Rights or other rights that might be claimed to
   pertain to the implementation or use of the technology described in
   this document or the extent to which any license under such rights
   might or might not be available; nor does it represent that it has
   made any independent effort to identify any such rights.  Information
   on the procedures with respect to rights in RFC documents can be
   found in BCP 78 and BCP 79.

   Copies of IPR disclosures made to the IETF Secretariat and any
   assurances of licenses to be made available, or the result of an
   attempt made to obtain a general license or permission for the use of
   such proprietary rights by implementers or users of this
   specification can be obtained from the IETF on-line IPR repository at
   http://www.ietf.org/ipr.

   The IETF invites any interested party to bring to its attention any
   copyrights, patents or patent applications, or other proprietary
   rights that may cover technology that may be required to implement
   this standard.  Please address the information to the IETF at ietf-
   ipr@ietf.org.

Acknowledgement

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







A. van Wijk, et al.    Expires September 5 2006              [Page 31]


Html markup produced by rfcmarkup 1.108, available from http://tools.ietf.org/tools/rfcmarkup/