Internet Engineering Task Force SIPPING WG
Internet Draft G. Hellström (editor)
Document: <draft-manyfolks-sipping-ToIP-01.txt> R. R. Roy (editor)
Feb 2004 A. van Wijk (editor)
Expires: August 2004 Omnitor, AT&T,
Informational Viataal
Framework of requirements for real-time text conversation using SIP.
Status of this Memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026 [1].
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Abstract
This document provides the framework of requirements for text
conversation with real time character-by-character interactive
flow over the IP network using the Session Initiation Protocol.
The requirements for general real-time text-over-IP telephony,
point-to point and conference calls, transcoding, relay services,
user mobility, interworking between text-over-IP telephony and
existing text-telephony, and some special features including
instant messaging have been described.
Table of Contents
1. Introduction 3
2. Scope 3
3. Terminology 3
4. Definitions 4
5. Background and General Requirements 5
6. Features in Real-time Text-over-IP 5
7. Real-Time Multimedia Conversational Sessions using SIP 6
8. General Requirements for Real-Time Text-over-IP using SIP 8
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8.1 Pre-Call Requirements 8
8.2 Basic Point-to-Point Call Requirements 9
8.2.1 General Requirements 9
8.2.2 Session Setup 9
8.2.3 Addressing 10
8.2.4 Alerting 10
8.2.5 Call Negotiations 10
8.2.6 Answering 11
8.2.7 Session progress and status presentation 11
8.2.8 Actions During Calls 12
8.2.9 Additional session control 13
8.2.10 File storage 14
8.3 Conference Call Requirements 14
8.4 Transport 14
8.5 Character Set 14
8.6 Transcoding 15
8.7 Relay Services 15
8.8 Emergency services 16
8.9 User Mobility 16
8.10 Confidentiality and Security 16
8.11 Call Flows 17
8.11.1 Call Scenarios 17
8.11.2 Point-to-Point Call Flows 18
8.11.3 Conference Call Flows 18
9. Interworking Requirements for Text-over-IP 19
9.1 Real-Time Text-over-IP Interworking Gateway Services 19
9.2 Text-over-IP and PSTN/ISDN Text-Telephony 19
9.3 Text-over-IP and Cellular Wireless circuit switched Text-
Telephony 20
9.3.1 No-gain 20
9.3.2 Cellular Text Telephone Modem (CTM) 20
9.3.3 Baudot mode 21
9.3.4 Data channel mode 21
9.3.5 Common Text Gateway Functions 21
9.4 Text-over-IP and Cellular Wireless Text-over-IP 21
9.5 Instant Messaging Support 21
9.6 IP Telephony with Traditional RJ-11 Interfaces 22
9.7 Interworking Call Flows 23
9.8 Multi-functional gateways 24
9.9 Gateway Discovery 24
9.10 Text Gateway in the call Scenarios 25
9.10.1 IP terminal calling an analogue textphone (PSTN) 25
9.10.2 IP terminal calling a mobile text telephone (CTM) 25
9.10.3 IP terminal calling a mobile telephone (GPRS based) 25
9.10.4 IP terminal calling a mobile telephone(UMTS) 26
9.10.5 Analogue textphone (PSTN) user calling an IP terminal 26
9.10.6 Mobile text telephone (CTM) user calling an IP terminal 26
9.10.7 Mobile telephone user (GPRS) calling an IP terminal 26
9.10.8 Mobile telephone (UMTS) user calling an IP terminal 26
9.10.9 Voice over DSL user using an analogue text telephone. 27
9.10.10 VoIP user via a building telephone switch (at an apartment
building) owning an analogue text telephone. 27
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9.10.11 VoIP user via a gateway/box connected to his/her own
Broadband connection owning an analogue text telephone. 27
10. Terminal Features 27
10.1 Text input 28
10.2 Text presentation 28
10.3 Call control 29
10.4 Device control 30
10.5 Alerting 30
10.6 External interfaces 30
10.7 Power 31
11. Security Considerations 31
12. Authors Addresses 31
13. Full Copyright Statement 32
14. References 33
14.1 Normative 33
14.2 Informative 34
1. Introduction
Text-over-IP (ToIP) is becoming popular as a part of total
conversation among a range of users although this medium of
communications may be the most convenient to certain categories of
people (e.g., deaf, hard of hearing and speech-impaired
individuals). The Session Initiation Protocol (SIP) has become the
protocol of choice for control of Multimedia IP telephony and
Voice-over-IP (VoIP) communications. Naturally, it has become
essential to define the requirements for how ToIP can be used with
SIP to allow text conversations as an equivalent to voice. This
document defines the framework of requirements for using ToIP,
either by itself or as a part of total conversation using SIP for
session control.
2. Scope
The primary scope of this document is to define the requirements
for using ToIP with SIP, either stand-alone or as a part of a
total conversation approach. In general, the scope of the
requirements is:
a. Features in Real-Time ToIP
b. Real-time Multimedia Conversational Sessions using SIP
c. General Requirements for Real-Time ToIP using SIP
d. Interworking Requirements for ToIP
e. Text gateways in the different networks
The subsequent sections describe those requirements in detail.
3. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL"
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in this document are to be interpreted as described in RFC 2119
[2].
4. Definitions
Full duplex user information is sent independently in both
directions.
Half duplex user information can only be sent in one direction
at a time or, if an attempt to send information in both directions
is made, errors can be introduced into the user information.
TTY name for text telephone, often used in USA, see textphone.
Textphone text telephone. A terminal device that allow end-to-end
real time text communication. A variety of textphone protocols
exists world-wide, both in the PSTN and other networks. A
textphone can often be combined with a voice telephone, or include
voice communication functions for simultaneous or alternating use
of text and voice in a call.
Text telephony Analog textphone services
Text Relay Service - A third-party or intermediary that enables
communications between deaf, hard of hearing and speech-impaired
people, and voice telephone users by translating between voice and
text in a call.
Transcoding Services - Services of a third-party user agent (human
or automated) that transcodes one stream into another.
Total Conversation - A multimedia service offering real time
conversation in video, text and voice according to interoperable
standards. All media flow in real time. Further defined in ITU-T
F.703 Multimedia conversational services description.
Text gateway A multi functionalgateway that sits at the border
of a network able to transcode RFC 2793 Interactive text (ToIP)
into a different text medium and vice versa. E.g. ToIP into Boudot
and vice versa in the PSTN.
Acronyms:
2G Second generation cellular (mobile)
2.5G Enhanced second generation cellular (mobile)
3G Third generation cellular (mobile)
CDMA Code Division Multiple Access
CTM Cellular Text Telephone Modem
GSM Global System of Mobile Communication
ISDN Integrated Services Digital Network
ITU-T International Telecommunications Union Telecommunications
standardisation Sector
PSTN Public Switched Telephone Network
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SIP Session Initiation Protocol
TDD Telecommunication Device for the Deaf
TDMA Time Division Multiple Access
ToIP Text over Internet Protocol
UTF-8 Universal Transfer Format 8
5. Background and General Requirements
The main purpose of this document is to provide a set of
requirements for real-time text conversation over the IP network
using the Session Initiation Protocol (SIP) [3]. The overall
requirements described are such that the real-time text can be
expressed as a part of the session description as a part of the
total conversation like any other media. Participants can
negotiate all media including real-time text conversation[4, 5].
This is a highly desirable function for all IP telephony
users,irrespective of whether the users are or are not deaf, hard
of hearing, or speech impaired.
It is important to understand that real-time text conversations
are significantly different from other text based communications
like email or instant messaging. Real-time text conversations
deliver an equivalent mode to voice conversations by providing
transmission of text character by character as it is entered, so
that the conversation can be followed closely and immediate
interaction take place, therefore providing the same mode of
interaction as voice telephony does. Store-and-forward systems
like email or messaging on mobile networks or non-streaming
systems like instant messaging are unable to provide that
functionality.
One particular application where real-time text is absolutely
essential, is the use of relay services between conversational
modes, like between text and voice.
Direct text emergency service calls, where time and continuous
connection are of the essence, is another essential application.
6. Features in Real-time Text-over-IP
While real-time Text-over-IP will be used for a wide variety of
services, an important field of application will be to provide a
text equivalent to voice conversation, in particular for deaf,
hard of hearing and speech-impaired users.
As such, it is crucial that the conversational nature of this
service is maintained. Text based communications exist in a
variety of forms, some non-conversational (SMS, text paging, E-
mail, newsgroups, message boards, etc.), others conversational
(TTY/TDD, Textphone, etc).
Real-time Text-over-IP will sometimes be used in conjunction with
a relay service [I] to allow text users to communicate with voice
users. With relay services, it is crucial that text characters are
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sent as soon as possible after they are entered. While buffering
MAY be done to improve efficiency, the delays SHOULD be kept as
small as possible. In particular, buffering of whole lines of text
MUST NOT be used.
In order to make Real-Time Text-over-IP the equivalent of what
voice is to hearing people, it needs to offer equivalent features
in terms of conversation as voice communications provides to
hearing people. To achieve that, real-time Text-over-IP MUST:
a. Offer Real-Time presentation of the conversation. This means
that text MUST be sent as soon as available, or with very small
delays. The delay MUST not be longer than 500 milliseconds,
b. Provide simultaneous transmission in both directions,
c. Except for the case of interworking with other networks and
protocols (e.g. TTY on PSTN) allow users to interrupt/barge in at
any time in the conversation.
d. Except for the case of interworking with other networks and
protocols, Real-Time Text-over-IP MUST support a transmission rate
of at least 30 characters/second.
e. Support sending redundant data as described in RFC 2793 [5].
f. Be possible to merge with video transmission.
The end-to-end delay in transmission MUST be less than 2000
milliseconds.
Many users will want to use multiple modes of communication during
the conversation, either at the same time or by switching between
modes e.g. between real-time Text-over-IP and voice. Native real-
time Text-over-IP systems MUST support at least the alternate use
of modalities and MAY support simultaneous use of modalities.
When communicating via a gateway to other networks and protocols,
the system MUST completely support the functionality for
alternating or simultaneous modalities as offered by the gateway.
When voice is supported on the terminal, the terminal MUST provide
volume control.
7. Real-Time Multimedia Conversational Sessions using SIP
The Session Initiation Protocol (SIP) [3] provides mechanisms for
creating, modifying, and terminating sessions for real-time
conversation with one or more participants using any combination
of media: Text, Video and Audio. However, participants are allowed
to negotiate on a set of compatible media types (e.g., Text,
Video, Audio) with session descriptions used in SIP invitations.
The standardized T.140 real-time text conversation [4], in
addition to audio and video communications, will be valuable
services to many. Real-time text can be expressed as a part of the
session description in SIP and will be a useful subset of the
Total Conversation (e.g., Real-time text, Video and Audio).
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This specification describes the framework for using the T.140
text conversation in SIP as a part of the multimedia session
establishment in real-time over a SIP network.
The session establishment using SIP defines procedures for how
T.140 text conversation can be supported using a RTP payload
defined in RFC 2793 [5]. The performance characteristics of T.140
will be determined using RTCP.
The session will not only define procedures between the SIP
devices having text conversation capability, but will also define
how sessions in SIP can be established between the text
conversation and audio/video/text capable devices transparently.
If there is any incompatibility between the terminals, e.g. T.140
only and audio-only terminals, the necessary transcoding services
will need to be invoked. This important service feature invites a
variety of rich capabilities in the transcoding server. For
example, speech-to-text (STT), text-to-speech (TTS), text bridging
after conversion from speech, audio bridging after conversion from
text, and other services can also be provided by the transcoding
and/or translation server. The session description protocol (SDP)
[6] used in SIP to describe the session also needs to be capable
of expressing these attributes of the session (e.g., uniqueness in
media mapping for conversion from one media to another for each
communicating party).
Real-time texts can also be presented in conjunction with video.
Alerting for T.140 terminals needs to be provided. Users may set
up text conversation sessions using SIP from any location. In
addition, user privacy and security MUST be provided for text
conversation sessions at least equal to that for voice.
The transcoding/translation services can be invoked in SIP using
different session establishment models [7]: Third party call
control [8] and Conference Bridge model [9].
Both point-to-point and multipoint communication need to be
defined for the session establishment using T.140 text
conversation. In addition, the interworking between T.140 text
conversation and text telephony conversation [10] is needed.
The general requirements for real-time text conversation using SIP
can be described as follows:
a. Session setup, modification and teardown procedures for point-
to-point and multimedia calls
b. Registration procedures and address resolutions
c. Negotiation procedures for device capabilities
d. Discovery and invocation of transcoding/translation services
between the media in the call
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e. Different session establishment models for
transcoding/translation services invocation: Third party call
control and Conference bridge model
f. Uniqueness in media mapping to be used in the session for
conversion from one media to another by the
transcoding/translation server for each communicating party
g. Media bridging services for T.140 real-time text, audio, and
video for multipoint communications
h. Transparent session setup, modification, and teardown between
text conversation capable and voice/video capable devices
i. Conversations to be carried out using T.140-over-RTP and RTCP
will provide performance report for T.140
j. Altering capability using text conversation during the session
establishment
k. T.140 real-time text presentation mixing with voice and video
l. T.140 real-time text conversation sessions using SIP, allowing
users to move from one place to another
m. Users privacy and security for sessions setup, modification,
and teardown as well as for media transfer
n. Interoperability between T.140 conversations and text telephony
8. General Requirements for Real-Time Text-over-IP using SIP
The communications environments for ToIP using SIP to set up the
conversation in real-time may vary from a simple point-to-point
call to multipoint calls in addition to the fact that ToIP can be
used in combination with other media like audio and video. In
order to establish the session in real-time, the communicating
parties SHOULD be provided with experiences like those of normal
telephony call setup. There may also be some need for pre-call
setup e.g. storing registration information in the SIP registrar
to provide information about how a user can be contacted. This
will allow calls to be set up rapidly and with proper addressing.
Similarly, there are requirements that need to be satisfied during
call set up when another media is preferred by a user. For
instance, some users may prefer to use audio while others want to
use text as their preferred choice of conversational mode. In this
case, transcoding services will need to be invoked for text-to-
speech (TTS) and speech-to-text (STT). The requirements for
transcoding services need to be negotiated in real-time to set up
the session.
The subsequent subsections describe those requirements in great
detail.
8.1 Pre-Call Requirements
The desire of the users for using ToIP as a medium of
communications can be expressed during registration time. Two
situations need to be considered in the pre-call setup
environment:
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a. User Preferences: It MUST be possible for a user to indicate a
preference for ToIP by registering that preference in a SIP
server. If the user is called by other party, preferences can be
invoked by the SIP server to accept or reject the call based on
the rules defined by the user. If the rules require that a
transcoding server is needed, the call can be re-directed or
handled accordingly.
b. Server to support User Preferences: SIP servers MUST have the
capability to act on users preferences for ToIP, based on the
users preferences defined during the pre-call setup registration
time.
8.2 Basic Point-to-Point Call Requirements
The point-to-point call will take place between two parties. The
requirements are described in subsequent sub-sections. They assume
that one or both of the communicating parties will indicate ToIP
as the preferred medium for conversation using SIP in the session
setup.
8.2.1 General Requirements
The general requirements are that ToIP will be chosen from the
available media as the preferred means of communication for the
session. However, there may be a need to invoke some underlying
capabilities in some cases, for example, a transcoding server may
be invoked if one of the users want to use a communication medium
other than ToIP.
The following entities MAY need to be involved to facilitate the
session establishment using ToIP as another medium:
a. Caller Preferences: SIP headers (e.g., Contact) can be used to
show that ToIP is the medium of choice for communications.
b. Called Party Preferences: The called party being passive can
formulate a clear rule indicating how a call should be handled
either using ToIP as a preferred medium or not, and whether a
designated SIP proxy needs to handle this call or it is handled in
the SIP user agent (UA).
c. SIP Server support for User Preferences: SIP servers can also
handle the incoming calls in accordance to preferences expressed
for ToIP. The SIP Server can also enforce ToIP policy rules for
communications (e.g., use of the transcoding server for ToIP).
8.2.2 Session Setup
Users will set up a session by identifying the remote party or the
service they will want to connect to. However, conversations could
be started using a mode other than real-time Text-over-IP. For
instance, the conversation might be established using voice and
the user could elect to switch to text, or add text, during the
conversation. Systems supporting real-time Text-over-IP MUST allow
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users to select any of the supported conversation modes at any
time, including mid-conversation.
Systems SHOULD allow the user to specify a preferred mode of
communication, with the ability to fall back to alternatives that
the user has indicated are acceptable.
If the user requests simultaneous use of text and voice, and this
is not possible either because the system only supports alternate
modalities or because of resource management on the network, the
system MUST try to establish a text-only communication. and the
user MUST be informed of this change throughout the process,
either in text or in a combination of modalities that MUST include
text.
Session setup, especially through gateways to other networks, MAY
require the use of prefixes or the use of specially formatted
URLs.
This MUST be supported by the terminal.
8.2.3 Addressing
The SIP [3] addressing schemes MUST be used for all entities. For
example SIP URL and Tel URL will be used for caller, called party,
user devices, and servers (e.g., SIP server, Transcoding server).
The right to include a transforming or translating service MUST
NOT require user registration in any specific SIP registrar.
8.2.4 Alerting
Systems supporting real-time Text-over-IP MUST have an alerting
method (e.g., for incoming calls and messages) that can be used by
deaf and hard of hearing people or provide a range of alternative,
but equivalent, alerting methods that are suitable for all users,
regardless of their abilities and preferences.
It should be noted that general alerting systems exist, and one
common interface for triggering the alerting action is a contact
closure between two conductors.
Among the alerting options are alerting on the user equipment and
specific alerting user agents registered to the same registrar as
the main user agent.
If present, identification of the originating party (for example
in the form of a URL or CLI) MUST be clearly presented to the user
in a form suitable for the user BEFORE answering the request. When
the invitation to initiate a conversation involving real-time
Text-over-IP originates from a gateway, this MAY be signalled to
the user.
8.2.5 Call Negotiations
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The Session Description Protocol (SDP) used in SIP [3] provides
the capabilities to indicate ToIP as a media for the call setup.
RFC 2793 [5] provides the RTP payload type for support of ToIP
which can be indicated in the SDP as a part of SDP INVITE, OK and
SIP/200/ACK for media negotiations. In addition, SIPs
offer/answer model can also be used in conjunction with other
capabilities including the use of a transcoding server for
enhanced call negotiations [7,8,9].
8.2.6 Answering
Systems SHOULD provide a best-effort approach to answering
invitations for session set-up and users should be kept informed
at all times about the progress of session establishment. On all
systems that both inform users of session status and support real-
time Text-over-IP, this information MUST be available in text, and
may be provided in other visual media.
8.2.6.1 Auto-Answer
Systems for real-time Text-over-IP MAY support an auto-answer
function, equivalent to answering machines on telephony networks.
If an auto-answer function is supported, it MUST support at least
160 characters for the recorded message. It MUST support incoming
text message storage of a minimum of 16000 characters, although
systems MAY support much larger storage.
When the auto-answer function is activated, user alerting MUST
still take place. The user MUST be allowed to monitor the auto-
answer progress and MUST be allowed to intervene during any stage
of the auto-answer and take control of the session.
8.2.7 Session progress and status presentation
During a conversation that includes real-time Text-over-IP, status
and session progress information MUST be provided in text. That
information MUST be equivalent to session progress information
delivered in any other format, for example audio. Users MUST be
able to manage the session and perform all session control
functions based on the textual session progress information.
The user MUST be informed of any change in modalities.
Session progress information MUST use simple language as much as
possible so that it can be understood by as many users as
possible.
The use of jargon or ambiguous terminology SHOULD be avoided at
all times. It is RECOMMENDED to let text information be used
together with icons symbolising the items to be reported.
There MUST be a clear indication, both visually as well as audibly
whenever a session gets connected and disconnected. The user
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should never be in doubt as to what the status of the connection
is, even if he/she is not able to use audio feedback or vision.
8.2.8 Actions During Calls
Certain actions need to be performed for the ToIP conversation
during the call and these actions are describe briefly as follows:
a. Text transmission SHALL be done character by character as
entered, or in small groups transmitted so that no character is
delayed between entry and transmission by more than 300
milliseconds.
b. The text transmission SHALL allow a rate of at least 30
characters per second so that human typing speed as well as speech
to text methods of generating conversation text can be supported.
c. After text connection is established, the mean end-to-end delay
of characters SHALL be less than two seconds, measured between two
ToIP users. This requirement is valid as long as the text input
rate is lower or equal to the text reception and display rate.
d. The character corruption rate SHALL be less than 1% in
conditions where users experience the quality of voice
transmission to be low but useable. This is in accordance with
ITU-T F.700 Annex A.3 quality level T1.
e. When interoperability functions are invoked, there may be a
need for intermediate storage of characters before transmission to
a device receiving slower than the typing speed of the sender.
Such temporary storage SHALL be dimensioned to adjust for
receiving at 30 characters per second and transmitting at 6
characters per second during at least 4 minutes [less than 3k].
f. If text is detected to be missing after transmission, there
SHALL be an indication in the text marking the loss.
g. When used from a terminal designed for PSTN text telephony, or
in interworking with such a terminal, ToIP shall enable
alternating between text and voice in a similar manner as the PSTN
text telephone handles this mode of operation. (This mode is often
called VCO/HCO in USA).
h. The transmission of the text conversation SHALL be made
according to an internationally suitable character set and control
protocol for text conversation as specified in ITU-T T.140.
i. When display of the conversation on end user equipment is
included in the design, display of the dialogue SHALL be made so
that it is easy to read text belonging to each party in the
conversation.
8.2.8.1 Text and other Media Handling Between ToIP Devices
The native ToIP devices do not need transcoding from speech to
text and can communicate directly.
I. When used between terminals designed for native ToIP, it SHALL
be possible to send and receive text simultaneously with the other
media (text, audio and/or video) supported by the same terminals.
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II. When used between terminals designed for native ToIP, it SHALL
be possible to send and receive text simultaneously.
8.2.8.2 General Actions
a. It SHALL be possible to establish a session with text
capabilities enabled at the beginning of a Call. Note: a call is
in this document defined as one or more sessions).
b. It SHALL be possible to place a call without text capabilities,
and to add text capabilities later in the call.
c. It SHALL be possible to transfer text at at least 30 characters
per second
d. It SHALL be possible to talk and listen simultaneously with
typing and reading.
8.2.8.3 Call Action with Native ToIP Devices
a. It SHALL be possible to answer a callwith text capabilities
enabled.
b. It SHOULD be possible to use video simultaneously with the
other media in the call.
c. It SHALL be possible to answer a callin voice or video without
text enabled, and add text later in the call.
d. It SHALL be possible to disconnect the call.
e. It SHOULD be possible to control IVR (Interactive Voice
Response) services from a numeric keypad.
f. It SHOULD be possible to control ITR ( Interactive Text
Response) services from the alphanumeric keyboard.
g. It SHOULD be possible to invoke multi-party calls.
h. It SHALL be possible to transfer the call.
i. It SHOULD be possible to use text characters (numbers) instead
of DTMF tones (numbers) in interactions where the person is using
a keyboard to interact with a service and the service asks for a
number.
8.2.8.4 Audio/Visual/Tactile Indicators
It SHALL be possible to observe visual or tactile indicators
about:
- Call progress
- Availability of text, voice and video channels.
- Incoming call.
- Incoming text.
- Typed and transmitted text.
- Any loss in incoming text.
8.2.9 Additional session control
Systems that support additional session control features, for
example call waiting, forwarding, hold etc on voice calls, MUST
offer equivalent functionality for real-time Text-over-IP
functions. In addition, all these features MUST be controllable by
text users at any time, in an equivalent way as for other users.
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It SHOULD be possible to use text characters (numbers) instead of
DTMF tones (numbers) in interactions where the person is using a
keyboard to interact with a service and the service asks for a
number.
8.2.10 File storage
Systems that support real-time Text-over-IP MAY save the text
conversation to a file. This SHOULD be done using a standard file
format. It is recommended to use an xhtml [11] format.
8.3 Conference Call Requirements
The conference call requirements deal with multipoint conferencing
calls where there will be at least one or more ToIP capable
devices along with other end user devices where the total number
end user devices will be at least three.
8.4 Transport
ToIP SHALL use RTP as the default transport protocol for
transmission of real-time text as specified in RFC 2793 [5].
Signaling and other media will use the transport protocol
specified in SIP [3] and/or their revised versions as specified in
standards.
The redundancy method of RFC 2198 SHOULD be used for making text
transmission reliable with transmission of three generations.
Text capability SHOULD be announced in SDP by a declaration in
line with this example:
m=text 11000 RTP/AVP 98 100
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98
Characters SHOULD BE buffered for transmission and transmitted
every 300 ms.
By having this single coding and transmission scheme for real time
text defined, in the SIP call control environment, the opportunity
for interoperability is optimised.
However, if good reasons exist, other transport mechanisms MAY be
offered and used for the T.140 coded text, provided that proper
negotiation is introduced, and RFC 2793 transport is used as the
defaut fallback solution.
8.5 Character Set
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a. Real-Time Text-over-IP protocols MUST use UTF-8 encoding as
specified in ITU-T T.140 [12]. A number of characters used in
traditional text telephony have special meanings.
b. Real-time Text-over-IP SHALL handle characers with editing
effect such as new line, erasure and alerting during session as
specified in ITU-T T.140.
8.6 Transcoding
Transcoding of text may need to take place in gateways between
ToIP and other forms of text conversation. ToIP make use of ISO
10646 character set.
Most PSTN textphones use a 7-bit character set, or a character set
that is converted to a 7-bit character set by the V.18 modem.
When transcoding between these character sets and T.140 in
gateways, special consideration MUST be paid to the national
variants of the 7 bit codes, with national characters mapping into
different codes in the ISO 10 646 code space. The national variant
to be used SHOULD be possible to select by the user per call, or
be configured as a national default for the gateway.
The missing text indicator in T.140, specified in T.140 amendment
1, cannot be represented in the 7 bit character codes. Therefore
these characters SHALL be translated to be represented by the '
(apostrophe) character in legacy text telephone systems where this
character exists. For legacy systems where the character ' does
not exist, the character . ( full stop ) SHALL be used instead.
8.7 Relay Services
The relay service acts as an intermediary between 2 or more
callers.
The basic relay service allows a translation of speech to text and
text to speech, which enables hearing and speech impaired callers
to communicate with hearing callers. Even though this document
focuses on ToIP, we do not exclude video relay services for e.g.,
speech to sign language and vice versa and other possible relay
services. It will be possible to use ToIP simultaneously with
other relay services if desired.
It is very important for the users that a relay session is invoked
as transparently as possible. It SHOULD happen automatically when
the call is being set-up or by a simple user action. A transcoding
framework document using SIP [7] describes invoking relay
services, where the relay acts as a conference bridge or uses the
third party control mechanism.
Adding or removing a relay service MUST be possible without
disrupting the current call.
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When setting up a call, the relay service MUST be able to
determine the type of service requested (e.g. speech to text or
text to speech), to indicate if the caller wants voice carry over,
the language of the text including the sign language being used.
The user MUST be provided with a method to indicate which service
is desired.
It MUST be possible to identify ToIP sessions as emergency
sessions.
The relay service operator MUST be able to process such a session
correctly and quickly.
a. The relay service operators network must give priority to this
incoming call.
b. The relay service operator MUST forward this session if they
are unable to process it to an alternative emergency relay
operator.
c. The relay service MUST label the transcoded stream as an
emergency call (in case of text to speech and/or vice versa).
d. The relay service MUST provide all session information to the
emergency centre (e.g., location information of the caller if
available).
Relay services must be available all the time, even if the users
are roaming.
8.8 Emergency services
a. It SHALL be possible to support emergency service calls with
text only or simultaneously with voice.
b. All session information that accompanies a voice session to the
emergency centre, shall also be provided to the emergency centre
if it is a ToIP session.(e.g, phone number and location
information of the user placing the emergency call).
c. A text over IP stream must be labelled as an emergency stream
to ensure that the emergency service center is able to receive
this call.
8.9 User Mobility
ToIP terminals SHALL use the same mechanisms as other terminals to
resolve mobility issues. It is RECOMMENDED to use a SIP-adress for
the users, resolved by a SIP REGISTRAR, to enable basic user
mobility. Further mechanisms are defined for the 3G IP multimedia
systems.
8.10 Confidentiality and Security
Users confidentiality and privacy need to be met as described in
SIP [3]. For example, nothing should reveal the fact that the user
of ToIP is a person with a disability unless the user prefers to
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make this information public. If a transcoding server is being
used, this SHOULD be transparent. Encryption SHOULD be used on
end-to-end or hop-by-hop basis as described in SIP [3].
Authentication needs to be provided for users in addition to the
message integrity and access control.
Protection against Denial-of-service (DoS) attacks needs to be
provided considering the case that the ToIP users might need
transcoding servers.
8.11 Call Flows
ToIP is a way of establishing the real-time conversation. Call
flow for ToIP SHOULD be as similar to audio and video session
establishment. For example, ToIP services MAY be invoked in the
following situations (among others):
- Noisy environment (e.g., in a machine room of a factory where
listening is difficult)Busy with another call and want to
participate in two calls at the same time
- Text and/or speech recording services (e.g., text
documentation/audio recording for legal/clarity/flexibility
purposes)
- Overcoming of language barriers through speech translation
and/or transcoding services
- Not hearing well or at all (e.g., hearing loss due to aging,
heard of hearing, deaf)
NOTE: In many of the above scenarios, text may accompany speech in
a caption like fashion. This would occur for individuals who are
hard of hearing and also for mixed calls with a hearing and deaf
person listening to the call.
All call flows either for the point-to-point or for the multipoint
need to consider that ToIP services may be invoked for many
different reasons by users as explained. When the
transcoding/translation services are needed, call flows will be
shown for both session establishment models: Third-party call
control model and Conferencing bridge model.
8.11.1 Call Scenarios
There are 2 different terminal types possible:
1. The terminal itself has the intelligence to initiate a relay
service for incoming and outgoing calls (based on address book,
user preferences programmed on the terminal etc. This terminal can
be used in a conference bridge call as well as a third party
control call.
2. Dumb terminals, so that the relay service server actually
initiates the correct call handling (the dumb terminal can only
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REFER the call to the relay center, which then sets up the call
using the conference bridge flow.).
The following call scenarios are shown:
- Communications between two ToIP/Multimedia capable, end user
devices using the same language.
- Communications between ToIP capable, end user devices using
translation services to provide language translation.
- Communications between ToIP/Multimedia capable and Audio (non-
ToIP) capable end user devices.
- Communications between ToIP/Multimedia and/or Audio (non-
ToIP)/Multimedia end user devices maintaining privacy.
8.11.2 Point-to-Point Call Flows
The point-to-point calls will contain at least one or both
ToIP/Multimedia devices in setting up the session. The detail call
flows need to be provided in the following scenarios:
- ToIP/Multimedia devices that use the same language.
- ToIP/Multimedia devices invoke translation services for using
different languages.
* Third-party call control model.
* Conference bridge service model.
- ToIP/Multimedia devices invoke translation services for using
different languages maintaining privacy.
* Third-party call control model.
* Conference bridge service model.
- ToIP/Multimedia device and Audio (non-ToIP)/Multimedia device
invoking transcoding server.
* Call initiated by Audio (non-ToIP)/Multimedia user
- Third-party call control model.
- Conference bridge service model.
* Call initiated by ToIP user.
- Third-party call control model.
- Conference bridge service model.
- ToIP/Multimedia device and Audio (non-ToIP)/Multimedia device
invoking transcoding server maintaining privacy.
* Call initiated by Audio (non-ToIP)/Multimedia user
- Third-party call control model.
- Conference bridge service model.
* Call initiated by ToIP user.
- Third-party call control model.
- Conference bridge service model.
8.11.3 Conference Call Flows
Conference call flows only contain the multipoint communications
scenarios, and only the centralized bridge model is considered.
The following multipoint conference call flow scenarios will
contain at least one more ToIP/Multimedia devices:
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- ToIP/Multimedia devices that use the same language.
- ToIP/Multimedia devices invoke translation services for using
different languages.
- ToIP/Multimedia devices invoke translation services for using
different languages maintaining privacy.
- ToIP/Multimedia device and Audio (non-ToIP)/Multimedia device
invoking transcoding server.
* Call initiated by Audio (non-ToIP)/Multimedia user.
* Call initiated by ToIP/Multimedia user.
- ToIP/Multimedia device and Audio (non-ToIP)/Multimedia device
invoking transcoding server maintaining privacy.
* Call initiated by Audio (non-ToIP)/Multimedia user.
* Call initiated by ToIP/Multimedia user.
9. Interworking Requirements for Text-over-IP
A number of systems for real time text conversation already exist
as well as a number of message oriented text communication
systems. Interoperability is of interest between ToIP and some of
these systems. This section describes requirements on this
interoperability.
9.1 Real-Time Text-over-IP Interworking Gateway Services
Interactive texting facilities exist already in various forms and
on various networks. On the PSTN, it is commonly referred to as
text telephony. The simultaneous or alternating use of voice and
text is used by a large number of users who can send voice, but
must receive text or who can hear but must send text due to a
speech disability.
9.2 Text-over-IP and PSTN/ISDN Text-Telephony
On PSTN networks, transmission of interactive text takes place
using a variety of codings and modulations, including ITU-T V.21
[II], Baudot, DTMF, V.23 [III] and others. Many difficulties have
arisen as a result of this variety in text telephony protocols and
the ITU-T V.18 [10] standard was developed to address some of
these issues.
ITU-T-V.18 [10] offers a native text telephony method plus it
defines interworking with current protocols. In the interworking
mode, it will recognise one of the older protocols and fall back
to that transmission method when required.
In order to allow systems and services based on Real-time Text-
over-IP to communicate with PSTN text telephones, text gateways
are the recommended approach. These gateways MUST use the ITU-T
V.18 [10] standard at the PSTN side.
Buffering MUST be used to support different transmission rates. At
least 1K buffer MUST be provided. 2K is recommended. In addition,
the gateway MUST provide a minimum throughput of at least 30
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characters/second or the highest speed supported by the PSTN text
telephony protocol side, whichever is the lowest.
PSTN-Real-time Text-over-IP gateways MUST allow alternating use of
text and voice.
PSTN and ISDN to real-time Text-over-IP gateways that receive CLI
information from the originating party MUST pass this information
to the receiving party as soon as possible.
Priority MUST be given to calls labeled as emergency calls.
9.3 Text-over-IP and Cellular Wireless circuit switched Text-
Telephony
Cellular wireless (or Mobile) circuit switched connections provide
a digital real-time transport service for voice or data.
The access technologies include GSM, CDMA, TDMA, iDen and various
3G technologies.
Alternative means of transferring the Text telephony data have
been developed when TTY services over cellular was mandated by the
FCC in the USA. They are a) No-gain codec solution, b) the
Cellular Text Telephony Modem (CTM) solution and c) Baudot mode
solution.
The GSM and 3G standards from 3GPP make use of the CTM modem in
the voice channel for text telephony.
However, implementations also exist that use the data channel to
provide such functionality. Interworking with these solutions
SHOULD be done using text gateways that set up the data channel
connection at the GSM side and provide real-time Text-over-IP at
the other side.
9.3.1 No-gain
The No-gain text telephone transporting technology uses
specially modified EFR [15] and EVR [16] speech vocoders in both
mobile terminals used provide a text telephony call. It provides
full duplex operation and supports alternating voice and
text.("VCO/HCO").
9.3.2 Cellular Text Telephone Modem (CTM)
CTM [17] is a technology independent modem technology that
provides the transport of text telephone characters at up to 10
characters/sec using modem signals that are at or below 1 kHz and
uses a highly redundant encoding technique to overcome the fading
and cell changing losses. On any interface that uses analog
transmission, half-duplex operation must be supported as the
send and receive modem frequencies are identical. The use of
CTM may have to be modified slightly to support half-duplex
operation.
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9.3.3 Baudot mode
This term is often used by cellular terminal suppliers for a GSM
cellular phone mode that allows TTYs to operate into a cellular
phone and to communicate with a fixed line TTY.
9.3.4 Data channel mode
Many mobile terminals allow the use of the data channel to
transfer data in real-time. Data rates of 9600 bit/s are usually
supported.
9.3.5 Common Text Gateway Functions
Text Gateways MUST support the differences that result from
different text protocols. The protocols to be supported will
depend on the service requirements of the Gateway.
Different data rates of different protocols MAY require text
buffering.
Interoperation of half-duplex and full-duplex protocols MAY
require text buffering and some intelligence to determine when to
change direction when operating in half-duplex.
Identification may be required of half-duplex operation either at
the user level (ie. users must inform each other) or at the
protocol level (where an indication must be sent back to the
Gateway).
A Text Gateway MUST be able to route text calls to emergency
service providers when any of the recognised emergency numbers
that support text communications for the country are called eg.
911 in USA.
A text gateway (MUST)/SHOULD act transparantly on the IP side. It
acts then as a virtual end-point terminal.
9.4 Text-over-IP and Cellular Wireless Text-over-IP
Text-over-IP MAY be supported over the cellular wireless packet
switched service. It interfaces to the Internet.
A Text gateway with cellular wireless packet switched services
MUST be able to route text calls into emergency service providers
when any of the recognized emergency numbers that support text
communication for the country are called.
9.5 Instant Messaging Support
Instant Messaging is used by many people to communicate using text
via the Internet. Instant Messaging transfers blocks of text
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rather than streaming as is used for real-time Text-over-IP. As
such, it is not a replacement for real-time Text-over-IP and in
particular does not meet the needs for real time conversations of
deaf, hard of hearing and speech-impaired users. It is unsuitable
for communications through a relay service [I]. The streaming
character of real-time Text-over-IP provides a better user
experience and, when given the choice, users often prefer real-
time Text-over-IP.
However, since some users might only have Instant Messaging
available, text gateways might be developed that allow
interworking between Instant Messaging systems and real-time Text-
over-IP solutions.
Because Instant Messaging is based on blocks of text, rather than
on a continuous stream of characters, such gateways need to
transform between these two formats. Text gateways for
interworking between Instant Messaging and real-time Text-over-IP
MUST concatenate individualcharacters originating at the real-time
Text-over-IP side into blocks of text and:
a. When the length of the concatenated message becomes longer than
50 characters, the buffered text MUST be transmitted to the
Instant Messaging side as soon as any non-alphanumerical character
is received from the real-time Text-over-IP side.
b. When a single carriage return, a single line feed, a carriage
return/line feed pair or a line feed/carriage return pair is
received from the real-time Text-over-IP side, the buffered
characters up to that point, including the carriage return and/or
line feed characters, MUST be transmitted to the Instant Messaging
side.
c. When the real-time Text-over-IP side has been idle for at least
5 seconds, all buffered text up to that point MUST be transmitted
to the Instant Messaging side.
Many Instant Messaging protocols signal that a user is typing to
the other party in the conversation. Text gateways between Instant
Messaging and real-time Text-over-IP MAY provide this signaling to
the Instant Messaging side when characters start being received,
either at the beginning of the conversation.
It is also possible to introduce the chat feature of certain
Instant Messaging protocols. When the chat feature is selected,
the IM client should use real-time text over IP. In this way, an
IM client can also be used for real-time streaming text over IP.
9.6 IP Telephony with Traditional RJ-11 Interfaces
Analogue adapters using SIP based IP communication and RJ-11
connectors for connecting traditional PSTN devices SHOULD enable
connection of legacy PSTN text telephones [18]. These adapters
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SHOULD contain V.18 modem functionality, voice handling
functionality, and conversion functions to/from SIP based ToIP
with T.140 transported in according to RFC 2793, in a similar way
as it provides interoperability for voice calls. If a call is set
up and RFC2793 capability is not declared by the endpoint (by the
end-point terminal or the text gateway in the network at the end-
point), a method for invoking a transcoding server shall be used.
If no such server is available, the signals from the textphone MAY
be transmitted in the voice channel as audio with high quality of
service.
NOTE: It is preferred that such analogue adaptors do use RFC2793
on board and thus act as a text gateway. Sending textphone signals
over the voice channel is undesirable due posible filtering and
compression between the 2 end-points. Which can result in dropping
characters in the textphone conversation or even not allowing the
textphones to connect with each other.
9.7 Interworking Call Flows
<< this chapter will change depending on how chapter 9.10 works
out>>
The call flows in chapter 8.11 deal with end to end ToIP. These
call flows do not change on the IP network when one end-point is
actually a text gateway. The text gateway actually acts like a
ToIP/Multimedia device. Separate call flows will show the
interworking between the ToIP/Multimedia devices [4] over the IP
network and the text telephony devices [10] over the PSTN/ISDN
network using the IP-PSTN/ISDN interworking functional (IWF)
entity. It is assumed that the IWF will provide ToIP and text
telephony interworking in addition to other capabilities. Thus
acting as a Text gateway.
The point-to-point call flows will contain at least one
ToIP/Multimedia and one text telephony/multimedia (or POTS) device
for the following cases:
- ToIP/Multimedia device and text telephony/multimedia device that
use the same/different language.
- ToIP/Multimedia device and PSTN/ISDN-based POTS/Multimedia
device.
For multipoint conferencing calls, it is assumed that only the
centralized conferencing will be considered, and the media bridge
is supposed to be located somewhere in the SIP network. However,
it is considered that the ToIP and text telephony interworking
function will be located in the IWF.
The multipoint conference call flows will contain at least one
ToIP/Multimedia, at least one text telephony/multimedia device,
and other devices where total number of devices will be three or
more for the following cases:
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- ToIP/Multimedia and text telephony/multimedia devices that use
the same/different language.
- ToIP/Multimedia devices, telephony/multimedia devices, and/or
PSTN/ISDN-based POTS/Multimedia devices.
9.8 Multi-functional gateways
The scenarios described in this document deal with single pairs of
interworking protocols or services. However, in practice many of
these interworking systems will be implemented as gateways that
combine different functions. As such, a text gateway could be
build to have modems to interwork with the PSTN and support both
Instant Messaging as well as real-time ToIP. Such interworking
functions are called Combination gateways.
Combination gateways MUST provide interworking between all of
their supported text based functions. For example, a text gateway
that has modems to interwork with the PSTN and that support both
Instant Messaging and real-time ToIP MUST support the following
interworking functions:
- PSTN text telephony to real-time ToIP.
- PSTN text telephony to Instant Messaging.
- Instant Messaging to real-time ToIP.
9.9 Gateway Discovery
To get a smooth invocation of the text gateways, where those
gateways are transparant on the IP side, it requires a method how
and when to invoke the text gateway. As described previously in
this draft. The text gateways must act as the end-terminal. The
capabilities of the text gateway will in that call be determined
by the call capabilities of the terminal that is using the
gateway. For example, a PSTN textphone is only able to receive
voice and streaming text. Thus the text gateway will only allow
ToIP and, in case of VCO or HCO, audio.
The PSTN devices or other non IP multimedia devices that require
the text gateways to connect to the IP must be able to locate the
text gateway. And ensure that the correct call capabilities of the
non IP multimedia device is used by the text gateway.
The following possible solutions for using the text gateway are:
- PSTN Textphone users using a prefix before dialing out.
- In band text dialogue. (???!!!)
- separate text subscriptions, linked to the phone number or
terminal identifier/ IP address.
- text capability indicators.
- text preference indicator.
- listen for text activity in all calls.
- call transfer request by the called user.
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- placing a call via the web.
- text gateways with its own telephone number and/or SIP address.
(this requires user interaction with the text gateway to place a
call).
- ENUM.
- etc
9.10 Text Gateway in the call Scenarios
9.10.1 IP terminal calling an analogue textphone (PSTN)
The ToIP stream will be converted into an analogue text telephone
protocol (using the voice channel) and vice versa by the text
gateway.
The PSTN knows it is a textphone call thanks to the SDP
description (for example: m=text 11000 RTP/AVP 98 a=rtpmap:98
t140/1000 for T.140 text on port 11000).
The PSTN will also know that all those incoming calls are only for
analogue textphones. Thus the speed of the text stream is adjusted
to the selected analogue textphone protocol.
If there is no analogue textphone on the called number, the call
setup will be terminated by the text gateway.
The text gateway can be implemented in 2 ways: The PSTN has its
own text gateway (the IWF), or it redirects the media stream to
the nearest IP-PSTN gateway with text transcoding abilities.
Text gateway detection: In the SIP messages.
9.10.2 IP terminal calling a mobile text telephone (CTM)
The ToIP stream will be converted into CTM and vice versa by the
text gateway located in the network of the cellular/mobile
operator. It is similar to the PSTN.
Text gateway detection: In the SIP messages.
9.10.3 IP terminal calling a mobile telephone (GPRS based)
A text gateway located in the mobile network converts the incoming
T.140/RTP stream into for example T.140 over TCP (T.140/TCP) or
tunnels the T.140 stream over HTTP (T.140/HTTP). Or any other
temporarily non standard solution necessary to connect the text
gateway with the text telephone client on the mobile phone.
This is necessary, since RTP over GPRS is not possible (especially
on GPRS phones with Symbian OS).
Note, those server-client solutions are ONLY acceptable for the
GPRS and non RTP stack phones. It is encouraged to use T.140/RTP
as soon as possible for all mobile phones.
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Allowing UDP transport over the GPRS link will enable RFC2793 text
over GPRS.
Text gateway detection: In the SIP messages.
9.10.4 IP terminal calling a mobile telephone(UMTS)
No text gateway is required here since this will be end to end IP.
9.10.5 Analogue textphone (PSTN) user calling an IP terminal
The PSTN is unable to distinguish between an analogue voice call
and an analogue textphone, both use the voice channel. The text
gateway needed to transcode the analogue textphone protocol into
T.140/RTP needs to be invoked.
The easiest way for a PSTN to separate an incoming voice call into
text telephony or normal voice is by using a prefix number for all
incoming text telephone calls to the PSTN. For example , the text
telephone user (e.g Boudot) places a call and enters a prefix e.g.
600 and then continues with the original number. The PSTN will
recognize all incoming 600 calls as an analogue textphone call and
redirects the call to a text gateway (unless it is a number
connecting the same PSTN).
It is undesirable to allow a PSTN to transport all the analogue
textphone tones/signals through a VoIP stream! (In band text
dialogue).
Text gateway detection: Prefix number for incoming textphone
calls.
9.10.6 Mobile text telephone (CTM) user calling an IP terminal
The voice channel of the cellular network is used. The MSC is able
to separate between the text call and voice only, it is just a
matter of redirecting the voice channel to the text gateway.
Text gateway detection: CTM signal detection.
9.10.7 Mobile telephone user (GPRS) calling an IP terminal
The text telephone client on the mobile telephone connects the
text gateway located in the network. The text gateway transcodes
the text stream into ToIP.
Text gateway detection: pre-programmed in the mobile textphone
client.
9.10.8 Mobile telephone (UMTS) user calling an IP terminal
No text gateway is required here since this will be end to end IP.
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9.10.9 Voice over DSL user using an analogue text telephone.
In Europe, Voice over DSL is introduced. It is likely that
analogue text telephones just use the voice channel. The VoDSL
gateway located in the network of the (A)DSL operator itself
should connect with a text gateway as soon it turns into VoIP.
Text gateway detection: prefix number similar to the PSTN.
9.10.10 VoIP user via a building telephone switch (at an apartment
building) owning an analogue text telephone.
This is the case where only VoIP is possible and no other IP
traffic between the telephone switch and the apartments. The
question is if this will be implemented.
The only solution would be a forced analogue text telephone
protocol over the Voice channel, in band text dialogue . If that
must happen. Then the telephone switch MUST convert the analogue
text telephone protocol into ToIP and vice versa before the
telephone switch connects the IP network.
Note: The in band text dialogue is undesirable. This scenario
SHOULD be avoided at any cost.
Text gateway detection: prefix number or in band text signalling.
9.10.11 VoIP user via a gateway/box connected to his/her own
Broadband connection owning an analogue text telephone.
The gateway box should natively transcode analogue text telephony
into ToIP and vice versa when an analogue text phone is plugged in
the RJ-11 socket [18].
Text gateway detection: RJ-11 socket preconfigured by the box via
jumpers or software.
10. Terminal Features
Implementers of products that support interactive Text-over-IP
SHOULD NOT assume that all users of text are able to use
mainstream input and output devices. People with arthritis or
other dexterity problems might not be able to use very small
keyboards. Visually impaired people might not be able to use
standard sized characters on a display. Colour-blind people might
suffer from badly chosen colour-schemes. People with motor
disabilities might require specialised input devices.
Implementers SHOULD try to make their products as open as possible
with regard to this wide range of abilities and preferences and
they MUST use standard interfaces wherever they provide such
interfaces.
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10.1 Text input
Systems that support real-time interactive Text-over-IP SHOULD
support suitable input mechanisms, either built-in or connectable
through the use of a standard interface: PS/2, USB, Bluetooth, or
virtual keyboard. In particular Braille users should be able to
connect Braille keyboards to the terminal. Terminals MAY support a
web interface for input and output of text.
It is recommended that systems that fixed terminals that support
real-time interactive Text-over-IP allow the user to enter the
standard alphanumerical characters directly, rather than through a
cycle of key presses or other indirect means. This could be done
using full-sized keyboards, smaller sized keyboards or fastap
keyboards for example. It is highly recommended to provide a
standard interface to allow attachment of an external input
device, especially for terminals that have only limited input
systems built-in.
All IP phones with a display of 12 or more characters MUST support
at least text input through the regular phone keypad (and display
of any incoming text) in order to provide basic emergency text
communication from any IP phone.
Input devices that have automatic key repeat MUST allow the user
to specify the key-repeat rate.
10.2 Text presentation
Systems that support real-time interactive Text-over-IP SHOULD
support suitable displays, either built-in or connectable through
the use of a standard interface: S-VGA, USB, Bluetooth or IP.
Braille readers should be connectable to the terminal using a
standard interface.
Terminals MAY support a web interface for input and output of
text.
While a variety of handsets and terminals might be developed for a
number of equally varied scenarios, implementers MUST:
In the case of fixed terminals or software applications on
Personal Computers:
a. Use either separate screen areas for displaying sent and
received text OR clearly indicate the difference between sent and
received text. Systems MAY allow the user to chose either on of
these presentation methodologies.
b. Provide at least 5 lines of 35 monospaced characters each for
each direction (sent and received text) OR at least 10 lines of 35
characters when sent and received text are presented together.
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In the case of Mobile terminals:
c. Use either separate screen areas for displaying sent and
received text OR clearly indicate the difference between sent and
received text. Systems MAY allow the user to chose either on of
these presentation methodologies.
d. Provide at least 3 lines of 20 monospaced characters each for
each direction (sent and received text) OR at least 6 lines of 20
characters when sent and received text are presented together.
On both types of terminals, scrolling back through both sent and
received text MUST be supported, including after the conversation
has ended. Lines SHOULD be wrapped at word boundaries and this is
strongly recommended.
There MUST be an easy-to-use function to clear the screen at any
time during the session, and if the implementation has chosen to
present sent and received text separately, clearing the screen
SHOULD be possible as a separate function for sent and received
text.
The function of the [CR], [LF] and [BACKSPACE] characters as
explained in section 9.5. MUST be supported by the presentation.
Presentation layers MUST support the full UTF-8 character set.
When real-time Text-over-IP is used in conjunction with other
modalities, like voice, the presentation MUST clearly indicate
this to the user in an area outside the display region for send
and received text.
Identification information for other parties in the conversation,
like URLs, user-friendly names from an address book, or CLI in
the case of conversations with text telephones, SHOULD be
displayed throughout the entire conversation in a region outside
the sent and received text area.
10.3 Call control
Call (Session) Control procedures MUST use the SIP protocol. Text
sessions MUST be identified in accordance with requirements
described earlier.
Text services SHOULD be part of a Total Conversation environment
in which voice, text and video sessions can be added, modified or
deleted individually.
To enable interworking with Textphones in telephone and cellular
(mobile) networks, terminals MUST be able to access Gateways
automatically when a PSTN or cellular (mobile) E.164-based
telephone number is used as the called address.
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Users MUST be able to establish text sessions to emergency service
providers using the widely recognised emergency numbers in use in
the country of operation of the terminal eg. 911 in USA.
The ability to transfer Location information SHALL be provided if
the information is available from the terminal.
10.4 Device control
ToIP will support the text protocol stack described earlier and
will require the use of RFC 2793 [5]. RFC 2793 defines the use of
ITU-T T.140 [4] over RTP. T.140 is a text presentation protocol
that is also used in the ITU-T H.series multimedia systems
including some videoconferencing systems. It is also used by ITU-T
V.18 [10], the Textphone interworking specification, and by the
GSM and 3GPP text conversation specifications.
ToIP will be a full-duplex service. Small displays may require the
users to indicate (via text indications at the user level) that a
user wishes to communicate in the half-duplex mode. This will
require a signal to inform the other user to proceed eg. GA as
traditionally used by many half-duplex TTY users.
10.5 Alerting
The form of Alerting indication(s) provided to the user should be
selectable to suit particular users. Alerting indications MAY
include Sound, Tactile (eg. vibrational), Visual (on-screen
symbols; separate flashing light), Motion (eg. movement of
something).
The ability to send an Alerting signal to an external interface
SHOULD be provided. This will allow Alerting devices that are
specific to users requirements to be attached.
As many as possible of the following alternatives for alerting
SHALL be provided:
* Internal flash.
* Two-pole connector for external alerting systems triggered
by contact between the two poles when a ring signal is generated.
* Bluetooth serial profile with AT command interface, sending
the "RING" message, intended for a Bluetooth alerting receiver
with flash, vibration or sound action.
* SIP connected alerting device, that get its stimuli by being
registered on the same sip address as the terminal.
10.6 External interfaces
Terminals for ToIP SHOULD provide external interfaces for the
following functions:
* Text input.
* Text display.
* Terminal control.
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* Session control.
10.7 Power
As terminals could remain active for very long periods of time,
the electrical power requirements of all the terminals SHOULD be
as low as possible.
If the terminal is to be used for calling Emergency services or
where the mains power supply is unreliable, back-up power systems
SHOULD be provided for the terminal and all equipment used to
provide the ToIP service. This can be implemented in many
different ways eg. via the line powering option on some Ethernet
interfaces, or by using a no break power supply (a battery back-
up system with inverters that can recreate a limited amount of
mains power).
11. Security Considerations
There are no additional security requirements other than described
earlier.
12. Authors Addresses
The following people provided substantial technical and writing
contributions to this document, listed alphabetically:
Barry Dingle
ACIF, 32 Walker Street
North Sydney, NSW 2060 Australia
Tel +61 (0)2 9959 9111
Fax +61 (0)2 9954 6136
TTY +61 (0)2 9923 1911
Mob +61 (0)41 911 7578
email barry.dingle@bigfoot.com.au
Guido Gybels
RNID, 19-23 Featherstone Street
London EC1Y 8SL, UK
Tel +44(0)20 7294 3713
Txt +44(0)20 7296 8019
Fax +44(0)20 7296 8069
EMail: guido.gybels@rnid.org.uk
Gunnar Hellstrom
Omnitor AB
Renathvagen 2
SE 121 37 Johanneshov
Sweden
Phone: +46 708 204 288 / +46 8 556 002 03
Fax: +46 8 556 002 06
Email: gunnar.hellstrom@omnitor.se
Paul E. Jones
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Cisco Systems, Inc.
7025 Kit Creek Rd.
Research Triangle Park, NC 27709
Phone: +1 919 392 6948
E-mail: paulej@packetizer.com
Radhika R. Roy
AT&T
Room C1-2B03
200 Laurel Avenue S.
Middletown, NJ 07748
USA
Phone: +1 732 420 1580
Fax: +1 732 368 1302
Email: rrroy@att.com
Henry Sinnreich
MCI
400 International Parkway
Richardson, Texas 75081
Email: henry.sinnreich@mci.com
Gregg C Vanderheiden
University of Wisconsin-Madison
Trace R & D Center
1550 Engineering Dr (Rm 2107)
Madison, Wi 53706
USA
gv@trace.wisc.edu
Phone +1 608 262-6966
FAX +1 608 262-8848
Arnoud A. T. van Wijk
Viataal (Dutch Institute for the Deaf)
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands.
Email: a.vwijk@viataal.nl
13. Full Copyright Statement
Copyright (C) The Internet Society (1999, 2000). All Rights
Reserved. This document and translations of it may be copied and
furnished to others, and derivative works that comment on or
otherwise explain it or assist in its implementation may be
prepared, copied, published and distributed, in whole or in part,
without restriction of any kind, provided that the above copyright
notice and this paragraph are included on all such copies and
derivative works. However, this document itself may not be
modified in any way, such as by removing the copyright notice or
references to the Internet Society or other Internet
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organizations, except as needed for the purpose of developing
Internet standards in which case the procedures for copyrights
defined in the Internet Standards process must be followed, or as
required to translate it into languages other than English.
The limited permissions granted above are perpetual and will not
be revoked by the Internet Society or its successors or assigns .
This document and the information contained herein is provided on
an "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET
ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
WARRANTIES OF MERCHANTABILITY OR FIT-NESS FOR A PARTICULAR
PURPOSE."
14. References
14.1 Normative
1. Bradner, S., "The Internet Standards Process -- Revision 3",
BCP 9, RFC 2026, October 1996.
2. Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997
3. J. Rosenberg, H. Schulzrinne, G. Camarillo, A. R. Johnston, J.
Peterson, R. Sparks, M. Handley, and E. Schooler, SIP: Session
Initiation Protocol, RFC 3621, IETF, June 2002.
4. ITU-T Recommendation T.140, Protocol for Multimedia
Application Text Conversation (February 1998) and Addendum 1
(February 2000).
5. G. Hellström, RTP Payload for Text Conversation, RFC 2793, May
2000.
6. G. Camarillo, H. Schulzrinne, and E. Burger, The Source and
Sink Attributes for the Session Description Protocol, IETF,
August 2003 - Work in Progress.
7. G.Camarillo,Framework for Transcoding with the Session
Initiation Protocol IETF august 2003 - Work in progress.
8. G. Camarillo, H. Schulzrinne, E. Burger, and A. Wijk,
Transcoding Service Invocation in SIP using Third Party Call
Control, IETF, August 2003 - Work in Progress.
9. G. Camarillo, The SIP Conference Bridge Transcoding Model,
IETF, August 2003 - Work in Progress.
10. ITU-T Recommendation V.18, Operational and Interworking
Requirements for DCEs operating in Text Telephone Mode, November
2000.
11. "XHTML 1.0: The Extensible HyperText Markup Language: A
Reformulation of HTML 4 in XML 1.0", W3C Recommendation. Available
at http://www.w3.org/TR/xhtml1.
12. Yergeau, F., "UTF-8, a transformation format of ISO 10646",
RFC 2279, January 1998.
13. TIA/EIA/825 A Frequency Shift Keyed Modem for Use on the
Public Switched Telephone Network. (The specification for 45.45
and 50 bit/s TTY modems.)
14. Bell-103 300 bit/s modem.
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15. TIA/EIA/IS-823-A TTY/TDD Extension to TIA/EIA-136-410
Enhanced Full Rate Speech Codec (must used in conjunction with
TIA/EIA/IS-840)
16. TIA/EIA/IS-127-2 Enhanced Variable Rate Codec, Speech
Service Option 3 for Wideband Spread Spectrum Digital Systems.
Addendum 2.
17. 3GPP TS26.226 Cellular Text Telephone Modem Description
(CTM).
18. I. Butcher, S. Lass, D. Petrie, H. Sinnreich, and C.
Stredicke, SIP Telephony Device Requirements, Configuration and
Data, IETF, February 2004- Work in Progress.
14.2 Informative
I. A relay service allows the users to transcode between different
modalities or languages. In the context of this document, relay
services will often refer to text relays that transcode text into
voice and vice-versa. See for example http://www.typetalk.org.
II. International Telecommunication Union (ITU), 300 bits per
second duplex modem standardized for use in the general switched
telephone network. ITU-T Recommendation V.21, November 1988.
III. International Telecommunication Union (ITU), 600/1200-baud
modem standardized for use in the general switched telephone
network. ITU-T Recommendation V.23, November 1988.
IV. Third Generation Partnership Project (3GPP), Technical
Specification Group Services and System Aspects; Cellular Text
Telephone Modem; General Description (Release 5). 3GPP TS 26.226
V5.0.0, March 2001"SIP Telephony Device Requirements,
Configuration and Data" by manyfolks,IETF, October 2003.
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