Internet Engineering Task Force G. Hellström, Editor
Internet Draft Omnitor
Document: draft-manyfolks-sipping-toip-00.txt R. R. Roy, Editor
AT&T
October 2003
Expires: April 2004
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.
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Table of Contents
1. Introduction...................................................4
2. Scope..........................................................4
3. Terminology....................................................4
4. Definitions....................................................4
5. Background and General Requirements............................5
6. Features in Real-time Text-over-IP.............................6
7. Real-Time Multimedia Conversational Sessions using SIP.........7
8. General Requirements for Real-Time Text-over-IP using SIP......9
8.1 Pre-Call Requirements.........................................9
8.2 Basic Point-to-Point Call Requirements.......................10
8.2.1 General Requirements.......................................10
8.2.2 Session Setup..............................................10
8.2.3 Addressing.................................................11
8.2.4 Alerting...................................................11
8.2.5 Call Negotiations..........................................11
8.2.6 Answering..................................................12
8.2.7 Session progress and status presentation...................12
8.2.8 Actions During Calls.......................................12
8.2.9 Additional session control.................................15
8.2.10 File storage..............................................15
8.3 Conference Call Requirements.................................15
8.4 Transport....................................................15
8.5 Character Set................................................16
8.6 Transcoding..................................................16
8.7 Relay Services...............................................16
8.8 Emergency services...........................................17
8.9 User Mobility................................................17
8.10 Confidentiality and Security................................18
8.11 Call Flows..................................................18
8.11.1 Call Scenarios............................................18
8.11.2 Point-to-Point Call Flows.................................19
8.11.3 Conference Call Flows.....................................20
9. Interworking Requirements for Text-over-IP....................20
9.1 Real-Time Text-over-IP Interworking Gateway Services.........20
9.2 Text-over-IP and PSTN/ISDN Text-Telephony....................20
9.3 Text-over-IP and Cellular Wireless circuit switched Text-
Telephony........................................................21
9.3.1 No-gain..................................................21
9.3.2 Cellular Text Telephone Modem (CTM)........................22
9.3.3 Baudot mode..............................................22
9.3.4 Data channel mode..........................................22
9.3.5 Common Gateway Functions...................................22
9.4 Text-over-IP and Cellular Wireless Text-over-IP..............22
9.5 Instant Messaging Support....................................23
9.6 IP Telephony with Traditional RJ-11 Interfaces...............24
9.7 Interworking Call Flows......................................24
9.8 Multi-functional gateways....................................25
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9.9 Gateway Discovery............................................25
10. Terminal Features............................................25
10.1 Text input..................................................25
10.2 Text presentation...........................................26
10.3 Call control................................................27
10.4 Device control..............................................28
10.5 Alerting....................................................28
10.6 External interfaces.........................................28
10.7 Power.......................................................29
11. Security Considerations......................................29
12. Issues to be Resolved........................................29
13. Authors Addresses...........................................29
14. Acknowledgments..............................................31
15. Full Copyright Statement.....................................31
16. References...................................................31
16.1 Normative...................................................31
16.2 Informative.................................................32
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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
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" 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
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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.
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
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.
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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
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.
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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).
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.
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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
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
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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:
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
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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 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.
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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
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
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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 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
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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
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other media (text, audio and/or video) supported by the same
terminals.
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. <<( a call is
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.
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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. 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.
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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
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. 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,
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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.
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 callswith 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.
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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 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 other forms of 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
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(In the scenarions, we need to keep in mind that there are 2
different possibilities, 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, and
dumb terminals, so that the relay service server actually initiates
the correct call handling (the dumb terminal may just forward the
call to the relay and the relay sets up the call (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
o Third-party call control model
o Conference bridge service model
. ToIP/Multimedia devices invoke translation services for using
different languages maintaining privacy
o Third-party call control model
o Conference bridge service model
. ToIP/Multimedia device and Audio (non-ToIP)/Multimedia device
invoking transcoding server
o Call initiated by Audio (non-ToIP)/Multimedia user
. Third-party call control model
. Conference bridge service model
o 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
o Call initiated by Audio (non-ToIP)/Multimedia user
. Third-party call control model
. Conference bridge service model
o Call initiated by ToIP/Multimedia user
. Third-party call control model
. Conference bridge service model
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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:
. 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
o Call initiated by Audio (non-ToIP)/Multimedia user
o Call initiated by ToIP/Multimedia user
. ToIP/Multimedia device and Audio (non-ToIP)/Multimedia device
invoking transcoding server maintaining privacy
o Call initiated by Audio (non-ToIP)/Multimedia user
o 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.
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In order to allow systems and services based on Real-time Text-over-
IP to communicate with PSTN text telephones, 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
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 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
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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.
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. D
9.3.5 Common Gateway Functions
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 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.
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.
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A 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 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, 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. Gateways for interworking between Instant
Messaging and real-time Text-over-IP MUST concatenate individual
characters 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. Gateways between Instant Messaging
and real-time Text-over-IP MAY provide this signaling to the Instant
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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. These adapters 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, 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.
9.7 Interworking Call Flows
The interworking call flows will include the interworking scenarios
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.
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 gateway could be build to
have modems to interwork with the PSTN and support both Instant
Messaging as well as real-time Text-over-IP. Such interworking
functions are called Combination gateways.
Combination gateways MUST provide interworking between all of their
supported text based functions. For example, a gateway that has
modems to interwork with the PSTN and that support both Instant
Messaging and real-time Text-over-IP MUST support the following
interworking functions:
PSTN text telephony to real-time Text-over-IP
PSTN text telephony to Instant Messaging
Instant Messaging to real-time Text-over-IP
9.9 Gateway Discovery
TBD
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.
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
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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.
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.
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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.
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.
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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:
o Internal flash.
o Two-pole connector for external alerting systems triggered by
contact between the two poles when a ring signal is generated.
o Bluetooth serial profile with AT command interface, sending the
"RING" message, intended for a Bluetooth alerting receiver with
flash, vibration or sound action.
o 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:
o Text input
o Text display
o Terminal control
o Session control
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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. Issues to be Resolved
T.140 over TCP/IP as an alternative; possible benefits and
procedures.
TBD
13. 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
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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
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
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Framework requirements for Text over IP October 2003
5271 GD Sint-Michielsgestel
The Netherlands.
Email: a.vwijk@viataal.nl
14. Acknowledgments
15. 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 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."
16. References
16.1 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.
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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 ??
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)
16.2 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.
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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 manifolks,
IETF, October 2003
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