draft-ietf-sipping-toip-02.txt   draft-ietf-sipping-toip-03.txt 
SIPPING Workgroup A. van Wijk (editor) SIPPING Workgroup A. van Wijk (editor)
Internet-Draft Viataal Internet-Draft Viataal
Category: Informational Category: Informational
Expires: February 21 2006 August 22 2005 Expires: March 6 2006 September 7 2005
Framework of requirements for real-time text conversation using SIP Framework of requirements for real-time text conversation using SIP
draft-ietf-sipping-toip-02.txt draft-ietf-sipping-toip-03.txt
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[1]. [1].
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Copyright Notice Copyright Notice
Copyright (C) The Internet Society (2005). Copyright (C) The Internet Society (2005).
Abstract Abstract
This document provides the framework of requirements for real-time This document provides the framework of requirements for real-time
character-by-character interactive text conversation over the IP character-by-character interactive text conversation over the IP
network using the Session Initiation Protocol and the Real-Time network using the Session Initiation Protocol and the Real-Time
Transport Protocol. It discusses requirements for real-time Text- Transport Protocol. It discusses requirements for real-time Text-
over-IP as well as interworking between Text-over-IP and existing over-IP as well as interworking between Text-over-IP and existing
text telephony on the PSTN and other networks. text telephony on the PSTN and other networks.
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Table of Contents Table of Contents
1. Introduction.....................................................3 1. Introduction.....................................................3
2. Scope............................................................4 2. Scope............................................................4
3. Terminology......................................................4 3. Terminology......................................................4
4. Definitions......................................................4 4. Definitions......................................................4
5. Framework Description............................................6 5. Framework Description............................................6
5.1. General requirements for ToIP..................................6 5.1. General requirements for ToIP..................................6
5.1.1 General ToIP Summary..........................................8 5.1.1 General ToIP Summary..........................................8
5.2. General Requirements for ToIP Interworking.....................8 5.2. General Requirements for ToIP Interworking.....................8
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7.3 ToIP and Cellular Wireless ToIP................................19 7.3 ToIP and Cellular Wireless ToIP................................19
7.4 Instant Messaging Support......................................19 7.4 Instant Messaging Support......................................19
7.5 Common Text Gateway Functions..................................20 7.5 Common Text Gateway Functions..................................20
7.5.1 Protocol support.............................................20 7.5.1 Protocol support.............................................20
7.5.2 Relay buffer storage.........................................20 7.5.2 Relay buffer storage.........................................20
7.5.3 Emergency calls through gateways.............................21 7.5.3 Emergency calls through gateways.............................21
7.5.4 Text Gateway Invocation......................................21 7.5.4 Text Gateway Invocation......................................21
7.6 Home Gateways or Analog Terminal Adapters......................21 7.6 Home Gateways or Analog Terminal Adapters......................21
7.7 Multi-functional Combination gateways..........................22 7.7 Multi-functional Combination gateways..........................22
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7.8 Transcoding....................................................22 7.8 Transcoding....................................................22
7.9 Relay Services.................................................23 7.9 Relay Services.................................................23
7.9.1 Basic function of the relay service..........................23 7.9.1 Basic function of the relay service..........................23
7.9.2 Invocation of relay services.................................23 7.9.2 Invocation of relay services.................................23
8. Security Considerations.........................................23 8. Security Considerations.........................................23
9. Authors Addresses...............................................24 9. Authors Addresses...............................................24
10. References.....................................................25 10. References.....................................................25
10.1 Normative references..........................................25 10.1 Normative references..........................................25
10.2 Informative references........................................27 10.2 Informative references........................................27
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required for the ToIP framework. required for the ToIP framework.
The Real-Time Transport Protocol (RTP) is the protocol of choice The Real-Time Transport Protocol (RTP) is the protocol of choice
for real-time data transmission, and its use for interactive text for real-time data transmission, and its use for interactive text
payloads is described in RFC4103 [5]. payloads is described in RFC4103 [5].
This document defines a framework for ToIP to be used either by This document defines a framework for ToIP to be used either by
itself or as part of integrated, multi-media services, including itself or as part of integrated, multi-media services, including
Total Conversation. Total Conversation.
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2. Scope 2. Scope
This document defines a framework for the implementation of real- This document defines a framework for the implementation of real-
time ToIP, either stand-alone or as a part of multimedia services, time ToIP, either stand-alone or as a part of multimedia services,
including Total Conversation. It defines the: including Total Conversation. It defines the:
a. Requirements of Real-time, interactive text; a. Requirements of Real-time, interactive text;
b. Requirements for ToIP interworking; b. Requirements for ToIP interworking;
c. Description of ToIP using SIP and RTP; c. Description of ToIP using SIP and RTP;
d. Description of ToIP interworking with other text services. d. Description of ToIP interworking with other text services.
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alternating use of text and voice in a call. alternating use of text and voice in a call.
Text bridging - a function of a gateway service that enables the Text bridging - a function of a gateway service that enables the
flow of text through the service between the users involved in the flow of text through the service between the users involved in the
call. call.
Text gateway - a function that transcodes between different forms Text gateway - a function that transcodes between different forms
of text transport methods, e.g., between ToIP in IP networks and of text transport methods, e.g., between ToIP in IP networks and
Baudot or ITU-T V.21 text telephony in the PSTN. Baudot or ITU-T V.21 text telephony in the PSTN.
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Text Relay Service - a third-party or intermediary that enables Text Relay Service - a third-party or intermediary that enables
communications between deaf, hard of hearing and speech-impaired communications between deaf, hard of hearing and speech-impaired
people, and voice telephone users by translating between voice and people, and voice telephone users by translating between voice and
text in a call. text in a call.
Text telephony ū analog textphone service. Text telephony ū analog textphone service.
Total Conversation - a multimedia service offering real time Total Conversation - a multimedia service offering real time
conversation in video, text and voice according to interoperable conversation in video, text and voice according to interoperable
standards. All media flow in real time. (See ITU-T F.703 standards. All media flow in real time. (See ITU-T F.703
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SIP Session Initiation Protocol SIP Session Initiation Protocol
SRTP Secure Real Time Transport Protocol SRTP Secure Real Time Transport Protocol
TDD Telecommunication Device for the Deaf TDD Telecommunication Device for the Deaf
TDMA Time Division Multiple Access TDMA Time Division Multiple Access
TTY Analog textphone (Teletypewriter) TTY Analog textphone (Teletypewriter)
ToIP Text over Internet Protocol ToIP Text over Internet Protocol
UTF-8 Universal Transfer Format-8 UTF-8 Universal Transfer Format-8
VCO/HCO Voice Carry Over/Hearing Carry Over VCO/HCO Voice Carry Over/Hearing Carry Over
VoIP Voice over Internet Protocol VoIP Voice over Internet Protocol
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5. Framework Description 5. Framework Description
This framework defines the requirements of a text-based This framework defines the requirements of a text-based
conversational service that is the text equivalent of voice based conversational service that is the text equivalent of voice based
telephony. Real-time text conversation can be combined with other telephony. Real-time text conversation can be combined with other
conversational services like video or voice. conversational services like video or voice.
ToIP also offers an IP equivalent of analog text telephony ToIP also offers an IP equivalent of analog text telephony
services as used by deaf, hard of hearing and speech-impaired services as used by deaf, hard of hearing and speech-impaired
individuals. individuals.
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5.1. General requirements for ToIP 5.1. General requirements for ToIP
In order to make ToIP the text equivalent of voice services, it In order to make ToIP the text equivalent of voice services, it
needs to offer equivalent features in terms of conversationality needs to offer equivalent features in terms of conversationality
as voice telephony provides. To achieve that, ToIP needs to: as voice telephony provides. To achieve that, ToIP needs to:
a. Offer real-time presentation of the conversation; a. Offer real-time presentation of the conversation;
b. Provide simultaneous transmission in both directions; b. Provide simultaneous transmission in both directions;
c. Support both point-to-point and multipoint communication; c. Support both point-to-point and multipoint communication;
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d. Allow other media, like audio and video, to be used in d. Allow other media, like audio and video, to be used in
conjunction with ToIP; conjunction with ToIP;
e. Ensure that the text service is always available. e. Ensure that the text service is always available.
Real-time text is a useful subset of Total Conversation defined in Real-time text is a useful subset of Total Conversation defined in
ITU-T F.703 [23]. Users could use multiple modes of communication ITU-T F.703 [23]. Users could use multiple modes of communication
during the conversation, either at the same time or by switching during the conversation, either at the same time or by switching
between modes, e.g., between text and audio. between modes, e.g., between text and audio.
Users may invoke ToIP services for many different reasons: Users may invoke ToIP services for many different reasons:
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certain SIP call control functions are available for the audio certain SIP call control functions are available for the audio
media part of the session, these functions MUST also be supported media part of the session, these functions MUST also be supported
for the text media part of the same session. For example, call for the text media part of the same session. For example, call
transfer must act on all media in the session. transfer must act on all media in the session.
T.140 real-time text conversation [4], in addition to audio and T.140 real-time text conversation [4], in addition to audio and
video communications, is a valuable service for many users, video communications, is a valuable service for many users,
including those on non-IP networks. T.140 also provides for real- including those on non-IP networks. T.140 also provides for real-
time editing of the text. time editing of the text.
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5.1.1 General ToIP Summary 5.1.1 General ToIP Summary
The general requirements for ToIP are: The general requirements for ToIP are:
a. Session setup, modification and teardown procedures for point- a. Session setup, modification and teardown procedures for point-
to-point and multimedia calls to-point and multimedia calls
b. Registration procedures and address resolutions b. Registration procedures and address resolutions
c. Registration of user preferences c. Registration of user preferences
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Text gateways are used for converting between different media Text gateways are used for converting between different media
types. They could be used between networks or within networks types. They could be used between networks or within networks
where different transport technologies are used. where different transport technologies are used.
When communicating via a gateway to other networks and protocols, When communicating via a gateway to other networks and protocols,
the ToIP service SHOULD support the functionality for alternating the ToIP service SHOULD support the functionality for alternating
or simultaneous use of modalities as offered by the destination or simultaneous use of modalities as offered by the destination
network. network.
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Address information, both called and calling, SHOULD be Address information, both called and calling, SHOULD be
transferred, and possibly converted, when interworking between transferred, and possibly converted, when interworking between
different networks. different networks.
ToIP will often be used to access a relay service [I], allowing ToIP will often be used to access a relay service [I], allowing
text users to communicate with voice users. With relay services, text users to communicate with voice users. With relay services,
it is crucial that text characters are sent as soon as possible it is crucial that text characters are sent as soon as possible
after they are entered. While buffering may be done to improve after they are entered. While buffering may be done to improve
efficiency, the delays SHOULD be kept minimal. In particular, efficiency, the delays SHOULD be kept minimal. In particular,
buffering of whole lines of text will not meet character delay buffering of whole lines of text will not meet character delay
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devices devices
g. Buffering of text when interworking with media that transport g. Buffering of text when interworking with media that transport
text at different rates. text at different rates.
5.2.1 PSTN Interworking 5.2.1 PSTN Interworking
Analog text telephony is cumbersome because of incompatible Analog text telephony is cumbersome because of incompatible
national implementations where interworking was never considered. national implementations where interworking was never considered.
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A large number of these implementations have been documented in A large number of these implementations have been documented in
ITU-T V.18 [10], which also defines the modem detection sequences ITU-T V.18 [10], which also defines the modem detection sequences
for the different text protocols. The modem type identification for the different text protocols. The modem type identification
may in rare cases take considerable time depending on user may in rare cases take considerable time depending on user
actions. actions.
To resolve analog textphone incompatibilities, text telephone To resolve analog textphone incompatibilities, text telephone
gateways are needed to transcode incoming analog signals into gateways are needed to transcode incoming analog signals into
T.140 and vice versa. The modem capability exchange time can be T.140 and vice versa. The modem capability exchange time can be
reduced by the text telephone gateways initially assuming the reduced by the text telephone gateways initially assuming the
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and the US Baudot (i.e. 45 bit/s) type of text telephones. and the US Baudot (i.e. 45 bit/s) type of text telephones.
5.2.2.2 Cellular Text Telephone Modem (CTM) 5.2.2.2 Cellular Text Telephone Modem (CTM)
CTM [15] is a technology independent modem technology that CTM [15] is a technology independent modem technology that
provides the transport of text telephone characters at up to 10 provides the transport of text telephone characters at up to 10
characters/sec using modem signals that can be carried by many characters/sec using modem signals that can be carried by many
voice codecs and uses a highly redundant encoding technique to voice codecs and uses a highly redundant encoding technique to
overcome the fading and cell changing losses. overcome the fading and cell changing losses.
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5.2.2.3 Cellular "Baudot mode" 5.2.2.3 Cellular "Baudot mode"
This term is often used by cellular terminal suppliers for a GSM This term is often used by cellular terminal suppliers for a GSM
cellular phone mode that allows TTYs to operate into a cellular cellular phone mode that allows TTYs to operate into a cellular
phone and to communicate with a fixed line TTY. phone and to communicate with a fixed line TTY.
5.2.3 Cellular data channel mode 5.2.3 Cellular data channel mode
Many mobile terminals allow the use of the data channel to Many mobile terminals allow the use of the data channel to
transfer data in real-time. Data rates of 9600 bit/s are usually transfer data in real-time. Data rates of 9600 bit/s are usually
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information about how a user can be contacted. This will allow information about how a user can be contacted. This will allow
sessions to be set up rapidly and with proper routing and sessions to be set up rapidly and with proper routing and
addressing. addressing.
Similarly, there are requirements that need to be satisfied during Similarly, there are requirements that need to be satisfied during
session set up when other media are preferred by a user. For session set up when other media are preferred by a user. For
instance, some users may indicate their preferred modality to be instance, some users may indicate their preferred modality to be
audio while others may indicate text. In this case, transcoding audio while others may indicate text. In this case, transcoding
services might be needed for text-to-speech (TTS) and speech-to- services might be needed for text-to-speech (TTS) and speech-to-
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text (STT). The requirements for transcoding services need to be text (STT). The requirements for transcoding services need to be
negotiated in real-time to set up the session. negotiated in real-time to set up the session.
The subsequent subsections describe some of these requirements in The subsequent subsections describe some of these requirements in
detail. detail.
6.1. Pre-Session Requirements 6.1. Pre-Session Requirements
The need to use text as a medium of communications can be The need to use text as a medium of communications can be
expressed by users during registration time. Two situations need expressed by users during registration time. Two situations need
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6.2.2 Text transport 6.2.2 Text transport
A ToIP service MUST always support at least one Text media type. A ToIP service MUST always support at least one Text media type.
ToIP services MUST support the Real-Time Transport Protocol (RTP) ToIP services MUST support the Real-Time Transport Protocol (RTP)
[24] according to the specification of RFC4103 [5] for the [24] according to the specification of RFC4103 [5] for the
transport of text between participants. transport of text between participants.
RFC4103 describes the transmission of T.140 [4] on IP networks. RFC4103 describes the transmission of T.140 [4] on IP networks.
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6.2.3 Session Setup 6.2.3 Session Setup
Users will set up a session by identifying the remote party or the Users will set up a session by identifying the remote party or the
service they want to connect to. However, conversations could be service they want to connect to. However, conversations could be
started using a mode other than text. For instance, the started using a mode other than text. For instance, the
conversation might be established using audio and the user could conversation might be established using audio and the user could
subsequently elect to switch to text, or add text as an additional subsequently elect to switch to text, or add text as an additional
modality, during the conversation. Systems supporting ToIP MUST modality, during the conversation. Systems supporting ToIP MUST
allow users to select any of the supported conversation modes at allow users to select any of the supported conversation modes at
any time, including mid-conversation. any time, including mid-conversation.
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The SIP [3] addressing schemes MUST be used for all entities in a The SIP [3] addressing schemes MUST be used for all entities in a
ToIP session. For example, SIP URLĘs or Tel URLĘs are used for ToIP session. For example, SIP URLĘs or Tel URLĘs are used for
caller, called party, user devices, and servers (e.g., SIP server, caller, called party, user devices, and servers (e.g., SIP server,
Transcoding server). Transcoding server).
The right to include a transcoding service MUST NOT require user The right to include a transcoding service MUST NOT require user
registration in any specific SIP registrar, but MAY require registration in any specific SIP registrar, but MAY require
authorisation of the SIP registrar to invoke the service. authorisation of the SIP registrar to invoke the service.
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6.2.5 Alerting 6.2.5 Alerting
User Agents supporting ToIP MUST have an alerting method (e.g., User Agents supporting ToIP MUST have an alerting method (e.g.,
for incoming sessions) that can be used by deaf and hard of for incoming sessions) that can be used by deaf and hard of
hearing people or provide a range of alternative, but equivalent, hearing people or provide a range of alternative, but equivalent,
alerting methods that can be selected by all users, regardless of alerting methods that can be selected by all users, regardless of
their abilities. their abilities.
It should be noted that external alerting systems exist and one It should be noted that external alerting systems exist and one
common interface for triggering the alerting action is a contact common interface for triggering the alerting action is a contact
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For example, tactile indications could be used by deafblind For example, tactile indications could be used by deafblind
individuals. individuals.
In summary, it SHOULD be possible to observe indicators about: In summary, it SHOULD be possible to observe indicators about:
- Incoming session - Incoming session
- Availability of text, voice and video channels - Availability of text, voice and video channels
- Session progress - Session progress
- Incoming text - Incoming text
- Any loss in incoming text - Any loss in incoming text
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- Typed and transmitted text. - Typed and transmitted text.
For users who cannot use the audible alerter for incoming For users who cannot use the audible alerter for incoming
sessions, it is RECOMMENDED to include a tactile as well as a sessions, it is RECOMMENDED to include a tactile as well as a
visual indicator. visual indicator.
6.2.8 Session Negotiations 6.2.8 Session Negotiations
The Session Description Protocol (SDP) used in SIP [3] provides The Session Description Protocol (SDP) used in SIP [3] provides
the capabilities to indicate text as a medium in the session the capabilities to indicate text as a medium in the session
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6.2.10 Actions During a Session 6.2.10 Actions During a Session
Certain actions need to be performed during ToIP conversation: Certain actions need to be performed during ToIP conversation:
a. Text transmission from a terminal SHALL be performed character a. Text transmission from a terminal SHALL be performed character
by character as entered, or in small groups of characters, so by character as entered, or in small groups of characters, so
that no character is delayed from entry to transmission by more that no character is delayed from entry to transmission by more
than 300 milliseconds. than 300 milliseconds.
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b. The text transmission SHALL allow a rate of at least 30 b. The text transmission SHALL allow a rate of at least 30
characters per second so that human typing speed as well as characters per second so that human typing speed as well as
speech to text methods of generating conversation text can be speech to text methods of generating conversation text can be
supported. supported.
c. To enable the use of international character sets, the c. To enable the use of international character sets, the
transmission format for text conversation SHALL be UTF-8 [12], transmission format for text conversation SHALL be UTF-8 [12],
in accordance with ITU-T T.140. in accordance with ITU-T T.140.
d. If text is detected to be missing after transmission, there d. If text is detected to be missing after transmission, there
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default and the fallback transport. default and the fallback transport.
6.2.10.2 Handling Text and other Media. 6.2.10.2 Handling Text and other Media.
A call is one or more related sessions. The following requirements A call is one or more related sessions. The following requirements
apply to media handling during a call: apply to media handling during a call:
a. When used between User Agents designed for ToIP, it SHALL be a. When used between User Agents designed for ToIP, it SHALL be
possible to send and receive text simultaneously. possible to send and receive text simultaneously.
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b. When used between User Agents that support ToIP, it SHALL be b. When used between User Agents that support ToIP, it SHALL be
possible to send and receive text simultaneously with the other possible to send and receive text simultaneously with the other
media (text, audio and/or video) supported by the same media (text, audio and/or video) supported by the same
terminals. terminals.
c. It SHOULD be possible to know during a call that ToIP is c. It SHOULD be possible to know during a call that ToIP is
available, even if it is not invoked at call setup (e.g. when available, even if it is not invoked at call setup (e.g. when
only voice and/or video is used initially). To disable this, only voice and/or video is used initially). To disable this,
the user MUST disable the use of ToIP. This is possible during the user MUST disable the use of ToIP. This is possible during
registration at the SIP registrar. registration at the SIP registrar.
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alerting during a session as specified in ITU-T T.140. alerting during a session as specified in ITU-T T.140.
6.5 Emergency services 6.5 Emergency services
It MUST be possible to place an emergency call using ToIP and it 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 MUST be possible to use a relay service in such call. The
emergency service provided to users utilising the text medium MUST emergency service provided to users utilising the text medium MUST
be equivalent to the emergency service provided to users utilising be equivalent to the emergency service provided to users utilising
speech or other media. speech or other media.
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6.6 User Mobility 6.6 User Mobility
ToIP User Agents SHOULD use the same mechanisms as other SIP User ToIP User Agents SHOULD use the same mechanisms as other SIP User
Agents to resolve mobility issues. It is RECOMMENDED that users Agents to resolve mobility issues. It is RECOMMENDED that users
use a SIP-address, resolved by a SIP registrar, to enable basic use a SIP-address, resolved by a SIP registrar, to enable basic
user mobility. Further mechanisms are defined for all session user mobility. Further mechanisms are defined for all session
types for 3G IP multimedia systems. types for 3G IP multimedia systems.
6.7 Firewalls and NATs 6.7 Firewalls and NATs
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7.2 ToIP and PSTN/ISDN Text-Telephony Interworking 7.2 ToIP and PSTN/ISDN Text-Telephony Interworking
On PSTN networks, transmission of interactive text takes place On PSTN networks, transmission of interactive text takes place
using a variety of codings and modulations, including ITU-T V.21 using a variety of codings and modulations, including ITU-T V.21
[II], Baudot [III], DTMF, V.23 [IV] and others. Many difficulties [II], Baudot [III], DTMF, V.23 [IV] and others. Many difficulties
have arisen as a result of this variety in text telephony have arisen as a result of this variety in text telephony
protocols and the ITU-T V.18 [10] standard was developed to protocols and the ITU-T V.18 [10] standard was developed to
address some of these issues. address some of these issues.
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ITU-T V.18 [10] offers a native text telephony method plus it ITU-T V.18 [10] offers a native text telephony method plus it
defines interworking with current protocols. In the interworking defines interworking with current protocols. In the interworking
mode, it will recognise one of the older protocols and fall back mode, it will recognise one of the older protocols and fall back
to that transmission method when required. to that transmission method when required.
V.18 MUST be supported on the PSTN side of a PSTN-ToIP gateway. V.18 MUST be supported on the PSTN side of a PSTN-ToIP gateway.
PSTN-ToIP gateways MUST allow alternating use of text and voice if PSTN-ToIP gateways MUST allow alternating use of text and voice if
the PSTN textphone involved at the PSTN side of the session the PSTN textphone involved at the PSTN side of the session
supports this. (This mode is often called VCO/HCO). supports this. (This mode is often called VCO/HCO).
skipping to change at line 1016 skipping to change at line 1016
return and/or line feed characters, SHOULD be transmitted to return and/or line feed characters, SHOULD be transmitted to
the Instant Messaging side. the Instant Messaging side.
c. When the ToIP side has been idle for at least 5 seconds, all c. When the ToIP side has been idle for at least 5 seconds, all
buffered text up to that point SHOULD be transmitted to the buffered text up to that point SHOULD be transmitted to the
Instant Messaging side. Instant Messaging side.
It is RECOMMENDED that during the session, both users are It is RECOMMENDED that during the session, both users are
constantly updated on the progress of the text input. constantly updated on the progress of the text input.
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Many Instant Messaging protocols signal that a user is typing to Many Instant Messaging protocols signal that a user is typing to
the other party in the conversation. Text gateways between such the other party in the conversation. Text gateways between such
Instant Messaging protocols and ToIP MUST provide this signaling Instant Messaging protocols and ToIP MUST provide this signaling
to the Instant Messaging side when characters start being to the Instant Messaging side when characters start being
received, or at the beginning of the conversation. received, or at the beginning of the conversation.
At the ToIP side, an indicator of writing the Instant Message MUST At the ToIP side, an indicator of writing the Instant Message MUST
be present where the Instant Messaging protocol provides one. For be present where the Instant Messaging protocol provides one. For
example, the real-time text user MAY see ". . . waiting for example, the real-time text user MAY see ". . . waiting for
replying IM. . . " and when 5 seconds have passed another . (dot) replying IM. . . " and when 5 seconds have passed another . (dot)
skipping to change at line 1071 skipping to change at line 1071
receiving at 30 characters per second and transmitting at 6 receiving at 30 characters per second and transmitting at 6
characters per second for up to 4 minutes (i.e. less than 3k characters per second for up to 4 minutes (i.e. less than 3k
characters). characters).
Interoperation of half-duplex and full-duplex protocols MAY Interoperation of half-duplex and full-duplex protocols MAY
require text buffering. Some intelligence will be needed to require text buffering. Some intelligence will be needed to
determine when to change direction when operating in half-duplex determine when to change direction when operating in half-duplex
mode. Identification may be required of half-duplex operation mode. Identification may be required of half-duplex operation
either at the "user" level (ie. users must inform each other) or either at the "user" level (ie. users must inform each other) or
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at the "protocol" level (where an indication must be sent back to at the "protocol" level (where an indication must be sent back to
the Gateway). the Gateway).
7.5.3 Emergency calls through gateways 7.5.3 Emergency calls through gateways
A text gateway MUST be able to route text calls to emergency A text gateway MUST be able to route text calls to emergency
service providers when any of the recognised emergency numbers service providers when any of the recognised emergency numbers
that support text communications for the country or region are that support text communications for the country or region are
called e.g. "911" in USA and "112" in Europe. Routing text calls called e.g. "911" in USA and "112" in Europe. Routing text calls
to emergency services MAY require the use of a transcoding to emergency services MAY require the use of a transcoding
skipping to change at line 1126 skipping to change at line 1126
Analog terminal adapters (ATAs) using SIP based IP communication Analog terminal adapters (ATAs) using SIP based IP communication
and RJ-11 connectors for connecting traditional PSTN devices and RJ-11 connectors for connecting traditional PSTN devices
SHOULD enable connection of legacy PSTN text telephones [16]. SHOULD enable connection of legacy PSTN text telephones [16].
These adapters SHOULD contain V.18 modem functionality, voice These adapters SHOULD contain V.18 modem functionality, voice
handling functionality, and conversion functions to/from SIP based handling functionality, and conversion functions to/from SIP based
ToIP with T.140 transported according to RFC 4103 [5], in a ToIP with T.140 transported according to RFC 4103 [5], in a
similar way as it provides interoperability for voice sessions. similar way as it provides interoperability for voice sessions.
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If a session is set up and text/t140 capability is not declared by 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 the destination endpoint (by the end-point terminal or the text
gateway in the network at the end-point), a method for invoking a gateway in the network at the end-point), a method for invoking a
transcoding server SHALL be used. If no such server is available, transcoding server SHALL be used. If no such server is available,
the signals from the textphone MAY be transmitted in the voice the signals from the textphone MAY be transmitted in the voice
channel as audio with high quality of service. channel as audio with high quality of service.
NOTE: It is preferred that such analog terminal adaptors do use NOTE: It is preferred that such analog terminal adaptors do use
RFC 4103 [5] on board and thus act as a text gateway. Sending RFC 4103 [5] on board and thus act as a text gateway. Sending
textphone signals over the voice channel is undesirable due to textphone signals over the voice channel is undesirable due to
skipping to change at line 1180 skipping to change at line 1180
the 7 bit codes, with national characters mapping into different the 7 bit codes, with national characters mapping into different
codes in the ISO 10646 code space. The national variant to be used 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 could be selectable by the user on a per call basis, or be
configured as a national default for the gateway. configured as a national default for the gateway.
The indicator of missing text in T.140, specified in T.140 The indicator of missing text in T.140, specified in T.140
amendment 1, cannot be represented in the 7 bit character codes. amendment 1, cannot be represented in the 7 bit character codes.
Therefore the indicator of missing text SHOULD be transcoded to Therefore the indicator of missing text SHOULD be transcoded to
the ' (apostrophe) character in legacy text telephone systems, the ' (apostrophe) character in legacy text telephone systems,
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where this character exists. For legacy systems where the where this character exists. For legacy systems where the
character ' does not exist, the . ( full stop ) character SHOULD character ' does not exist, the . ( full stop ) character SHOULD
be used instead. be used instead.
7.9 Relay Services 7.9 Relay Services
The relay service acts as an intermediary between two or more The relay service acts as an intermediary between two or more
callers using different media or different media encoding schemes. callers using different media or different media encoding schemes.
7.9.1 Basic function of the relay service 7.9.1 Basic function of the relay service
skipping to change at line 1235 skipping to change at line 1235
guidelines for relay service invocation can be found in RFC 3351 guidelines for relay service invocation can be found in RFC 3351
[19]. [19].
8. Security Considerations 8. Security Considerations
User confidentiality and privacy need to be met as described in User confidentiality and privacy need to be met as described in
SIP [3]. For example, nothing should reveal the fact that the user 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 of ToIP is a person with a disability unless the user prefers to
make this information public. If a transcoding server is being make this information public. If a transcoding server is being
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used, this SHOULD be transparent. Encryption SHOULD be used on 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 end-to-end or hop-by-hop basis as described in SIP [3] and SRTP
[17]. [17].
Authentication needs to be provided for users in addition to the Authentication needs to be provided for users in addition to the
message integrity and access control. message integrity and access control.
Protection against Denial-of-service (DoS) attacks needs to be Protection against Denial-of-service (DoS) attacks needs to be
provided considering the case that the ToIP users might need provided considering the case that the ToIP users might need
transcoding servers. transcoding servers.
skipping to change at line 1286 skipping to change at line 1286
Gunnar Hellstrom Gunnar Hellstrom
Omnitor AB Omnitor AB
Renathvagen 2 Renathvagen 2
SE 121 37 Johanneshov SE 121 37 Johanneshov
Sweden Sweden
Phone: +46 708 204 288 / +46 8 556 002 03 Phone: +46 708 204 288 / +46 8 556 002 03
Fax: +46 8 556 002 06 Fax: +46 8 556 002 06
Email: gunnar.hellstrom@omnitor.se Email: gunnar.hellstrom@omnitor.se
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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 Henry Sinnreich
pulver.com pulver.com
115 Broadhollow Rd 115 Broadhollow Rd
Suite 225 Suite 225
Melville, NY 11747 Melville, NY 11747
USA USA
Tel: +1.631.961.8950 Tel: +1.631.961.8950
Gregg C Vanderheiden Gregg C Vanderheiden
University of Wisconsin-Madison University of Wisconsin-Madison
skipping to change at line 1331 skipping to change at line 1338
Levels", BCP 14, RFC 2119, IETF, March 1997 Levels", BCP 14, RFC 2119, IETF, March 1997
3. J. Rosenberg, H. Schulzrinne, G. Camarillo, A. R. Johnston, J. 3. J. Rosenberg, H. Schulzrinne, G. Camarillo, A. R. Johnston, J.
Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: Session Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: Session
Initiation Protocol", RFC 3621, IETF, June 2002. Initiation Protocol", RFC 3621, IETF, June 2002.
4. ITU-T Recommendation T.140, "Protocol for Multimedia 4. ITU-T Recommendation T.140, "Protocol for Multimedia
Application Text Conversation" (February 1998) and Addendum 1 Application Text Conversation" (February 1998) and Addendum 1
(February 2000). (February 2000).
A. van Wijk, et al. Expires 6 March 2006 [Page 25 of 28]
5. G. Hellstrom, "RTP Payload for Text Conversation", RFC 4103, 5. G. Hellstrom, "RTP Payload for Text Conversation", RFC 4103,
IETF, June 2005. IETF, June 2005.
6. G. Camarillo, H. Schulzrinne, and E. Burger, "The Source and 6. G. Camarillo, H. Schulzrinne, and E. Burger, "The Source and
Sink Attributes for the Session Description Protocol," IETF, Sink Attributes for the Session Description Protocol," IETF,
August 2003 - Work in Progress. August 2003 - Work in Progress.
A. van Wijk, et al. Expires 21 February 2006 [Page 25 of 28]
7. G.Camarillo, "Framework for Transcoding with the Session 7. G.Camarillo, "Framework for Transcoding with the Session
Initiation Protocol" IETF June 2005 - Work in progress. Initiation Protocol" IETF June 2005 - Work in progress.
8. G. Camarillo, H. Schulzrinne, E. Burger, and A. van Wijk, 8. G. Camarillo, H. Schulzrinne, E. Burger, and A. van Wijk,
"Transcoding Services Invocation in the Session Initiation "Transcoding Services Invocation in the Session Initiation
Protocol (SIP) Using Third Party Call Control (3pcc)" RFC 4117, Protocol (SIP) Using Third Party Call Control (3pcc)" RFC 4117,
IETF, June 2005. IETF, June 2005.
9. G. Camarillo, "The SIP Conference Bridge Transcoding Model," 9. G. Camarillo, "The SIP Conference Bridge Transcoding Model,"
IETF, August 2003 - Work in Progress. IETF, August 2003 - Work in Progress.
skipping to change at line 1383 skipping to change at line 1390
Device Requirements and Configuration," IETF, June 2005 - Work in Device Requirements and Configuration," IETF, June 2005 - Work in
Progress. Progress.
17. Baugher, McGrew, Carrara, Naslund, Norrman, "The Secure Real 17. Baugher, McGrew, Carrara, Naslund, Norrman, "The Secure Real
Time Transport Protocol (SRTP)", RFC 3711, IETF, March 2004. Time Transport Protocol (SRTP)", RFC 3711, IETF, March 2004.
18. "IP Multimedia default codecs". 3GPP TS 26.235 18. "IP Multimedia default codecs". 3GPP TS 26.235
19. Charlton, Gasson, Gybels, Spanner, van Wijk, "User 19. Charlton, Gasson, Gybels, Spanner, van Wijk, "User
Requirements for the Session Initiation Protocol (SIP) in Support Requirements for the Session Initiation Protocol (SIP) in Support
A. van Wijk, et al. Expires 6 March 2006 [Page 26 of 28]
of Deaf, Hard of Hearing and Speech-impaired Individuals", RFC of Deaf, Hard of Hearing and Speech-impaired Individuals", RFC
3351, IETF, August 2002. 3351, IETF, August 2002.
20. J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with the 20. J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with the
Session Description Protocol (SDP)", RFC 3624, IETF, June 2002. Session Description Protocol (SDP)", RFC 3624, IETF, June 2002.
21. ITU-T Recommendation F.700,"Framework Recommendation for 21. ITU-T Recommendation F.700,"Framework Recommendation for
Multimedia Services", November 2000. Multimedia Services", November 2000.
A. van Wijk, et al. Expires 21 February 2006 [Page 26 of 28]
22. H. Schulzrinne, S.Casner, R. Frederick, V. Jacobsone, "RTP: A 22. H. Schulzrinne, S.Casner, R. Frederick, V. Jacobsone, "RTP: A
Transport Protocol for Real-Time Applications", RFC 3550, IETF, Transport Protocol for Real-Time Applications", RFC 3550, IETF,
July 2003. July 2003.
23. ITU-T Recommendation F.703,"Multimedia Conversational 23. ITU-T Recommendation F.703,"Multimedia Conversational
Services", November 2000. Services", November 2000.
24. J. Rosenberg, H. Schulzrinne, P. Kyzivat, "Indicating User 24. J. Rosenberg, H. Schulzrinne, P. Kyzivat, "Indicating User
Agent Capabilities in the Session Initiation Protocol (SIP)", RFC Agent Capabilities in the Session Initiation Protocol (SIP)", RFC
3840, IETF, August 2004 3840, IETF, August 2004
skipping to change at line 1436 skipping to change at line 1444
Intellectual Property Rights or other rights that might be claimed Intellectual Property Rights or other rights that might be claimed
to pertain to the implementation or use of the technology to pertain to the implementation or use of the technology
described in this document or the extent to which any license described in this document or the extent to which any license
under such rights might or might not be available; nor does it under such rights might or might not be available; nor does it
represent that it has made any independent effort to identify any represent that it has made any independent effort to identify any
such rights. Information on the procedures with respect to rights such rights. Information on the procedures with respect to rights
in RFC documents can be found in BCP 78 and BCP 79. in RFC documents can be found in BCP 78 and BCP 79.
Copies of IPR disclosures made to the IETF Secretariat and any Copies of IPR disclosures made to the IETF Secretariat and any
assurances of licenses to be made available, or the result of an assurances of licenses to be made available, or the result of an
A. van Wijk, et al. Expires 6 March 2006 [Page 27 of 28]
attempt made to obtain a general license or permission for the use attempt made to obtain a general license or permission for the use
of such proprietary rights by implementers or users of this of such proprietary rights by implementers or users of this
specification can be obtained from the IETF on-line IPR repository specification can be obtained from the IETF on-line IPR repository
at http://www.ietf.org/ipr. at http://www.ietf.org/ipr.
The IETF invites any interested party to bring to its attention The IETF invites any interested party to bring to its attention
any copyrights, patents or patent applications, or other any copyrights, patents or patent applications, or other
proprietary rights that may cover technology that may be required proprietary rights that may cover technology that may be required
to implement this standard. Please address the information to the to implement this standard. Please address the information to the
IETF at ietf-ipr@ietf.org. IETF at ietf-ipr@ietf.org.
A. van Wijk, et al. Expires 21 February 2006 [Page 27 of 28]
Disclaimer of Validity Disclaimer of Validity
This document and the information contained herein are provided on This document and the information contained herein are provided on
an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE
REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND
THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES,
EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT
THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR
ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A
PARTICULAR PURPOSE. PARTICULAR PURPOSE.
skipping to change at line 1471 skipping to change at line 1479
Copyright (C) The Internet Society (2005). This document is Copyright (C) The Internet Society (2005). This document is
subject to the rights, licenses and restrictions contained in BCP subject to the rights, licenses and restrictions contained in BCP
78, and except as set forth therein, the authors retain all their 78, and except as set forth therein, the authors retain all their
rights. rights.
Acknowledgment Acknowledgment
Funding for the RFC Editor function is currently provided by the Funding for the RFC Editor function is currently provided by the
Internet Society. Internet Society.
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 End of changes. 

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