Internet Engineering Task Force SIPPING WG
Internet Draft
Document: <draft-ietf-sipping-toip-01.txt> A. van Wijk (editor)
July 18 2005 Viataal
Expires: January 17 2006
Informational
Framework of requirements for real-time text conversation using SIP.
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Copyright Notice
Copyright (C) The Internet Society (2005).
Abstract
This document provides the framework of requirements for real-time
character-by-character interactive text conversation over the IP
network using the Session Initiation Protocol and the Transport
Protocol for Real-Time Applications. It discusses requirements for
real-time Text-over-IP telephony as well as interworking between
Text-over-IP telephony and existing text telephony on the PSTN and
other networks.
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Table of Contents
1. Introduction 3
2. Scope 3
3. Terminology 3
4. Definitions 4
5. Framework Description 5
5.1. Background 5
5.2. Requirements for ToIP 6
5.3. Use of SIP and RTP 6
5.4. Requirements for ToIP Interworking 9
6. Detailed requirements for Text-over-IP 9
6.1. Pre-Call Requirements 10
6.2 Basic Point-to-Point Call Requirements 10
6.2.1 Session Setup 10
6.2.2 Addressing 11
6.2.3 Alerting and session progress presentation 11
6.2.4 Call Negotiations 12
6.2.5 Answering 12
6.2.6 Actions During Calls 13
6.2.7 Additional session control 14
6.2.8 File storage 15
6.3 Conference Call Requirements for ToIP User Agents 15
6.4 Transport via RTP 15
6.5 Character Set 16
6.6 Transcoding 16
6.7 Relay Services 16
6.8 Emergency services 17
6.9 User Mobility 17
6.10 Confidentiality and Security 17
7. Interworking Requirements for ToIP 17
7.1 ToIP Interworking Gateway Services 17
7.2 ToIP and PSTN/ISDN Text-Telephony 18
7.3 ToIP and Cellular Wireless circuit switched Text-Telephony
18
7.3.1 "No-gain" 19
7.3.2 Cellular Text Telephone Modem (CTM) 19
7.3.3 "Baudot mode" 19
7.3.4 Data channel mode 19
7.3.5 Common Text Gateway Functions 19
7.4 ToIP and Cellular Wireless ToIP 20
7.5 Instant Messaging Support 20
7.6 IP Telephony with Traditional RJ-11 Interfaces 21
7.7 Multi-functional gateways 22
7.8 ToIP interoperability with PSTN text telephones. 22
7.9 Gateway Discovery 22
8. Afterword 23
9. Security Considerations 23
10. Authors Addresses 24
11. References 25
11.1 Normative 25
11.2 Informative 27
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1. Introduction
For many years, text has been in use as a medium for
conversational, interactive dialogue between users in a similar
way as voice telephony is used. Such interactive text is different
from messaging and semi-interactive solutions like Instant
Messaging in that it offers an equivalent conversational
experience to users that cannot, or do not wish to, use voice. It
therefore meets a different set of requirements than other text-
based solutions already available on IP networks.
Traditionally, deaf, hard of hearing and speech-impaired people
are amongst the most proliferate users of conversational,
interactive text, but because of its interactivity, it is becoming
popular amongst mainstream user groups as well.
This document describes how existing IETF protocols can be used to
implement a Text-over-IP solution (ToIP). This ToIP framework is
specifically designed to be compatible with Voice-over-IP
environments, as well as meeting the userÆs requirements,
including those of deaf, hard of hearing and speech-impaired users
as described in RFC3351 [21].
The Session Initiation Protocol (SIP) is the protocol of choice
for control of Multimedia IP telephony and Voice-over-IP (VoIP)
communications. It offers all the necessary control and signaling
required for the ToIP framework.
The Real-Time Transport Protocol (RTP) is the protocol of choice
for real-time data transmission, and its use for interactive text
payloads is described in RFC4103 [5].
This document defines a framework for ToIP to be used either by
itself or as part of integrated services, including Total
Conversation.
2. Scope
The primary scope of this document is to define a framework for
the implementation of ToIP, either stand-alone or as a part of
wider services, including Total Conversation. In general, the
scope is:
a. Description of ToIP using SIP and RTP;
b. Requirements of Real-time, interactive text;
c. Requirements for ToIP interworking.
The subsequent sections describe those requirements in detail.
3. Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT
RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as
described in BCP 14, RFC 2119 [2] and indicate requirement levels
for compliant implementations.
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4. Definitions
Audio bridging - a function of a gateway or relay service that
enables an audio path through the service between the users
involved in the call.
Full duplex - media is sent independently in both directions.
Half duplex - media can only be sent in one direction at a time
or, if an attempt to send information in both directions is made,
errors can be introduced into the presented media.
Interactive text - a term for real time transmission of text in a
character-by-character fashion for use in conversational services,
often as a text equivalent to voice based conversational services.
TTY û alternative designation for a text telephone, often used in
USA, see textphone. Also called TDD, Telecommunication Device for
the Deaf.
Textphone û also ôtext telephoneö. A terminal device that allows
end-to-end real-time, interactive 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 bridging - a function of a gateway service that enables the
flow of text through the service between the users involved in the
call.
Text gateway - a multi functional gateway that is able to
transcode between different forms of text transport methods, e.g.,
between ToIP in IP networks and Baudot text telephony in the PSTN.
Text telephony û 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 that
transcodes one stream into another. Transcoding can be done by
human operators, in automated manner or a combination of both
methods. Text Relay Services are examples of a transcoding service
between text and audio.
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.
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Video Relay Service - A service that enables communications
between deaf and hard of hearing people, and hearing persons with
voice telephones by translating between sign language and spoken
language in a call.
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. Framework Description
5.1. Background
The main purpose of this document is to provide a framework
description for the implementation of real-time, interactive text
based conversational services over IP networks, known as Text-
over-IP (ToIP).
This framework uses existing standards that are already commonly
used for voice based conversational services on IP networks. In
particular, the ToIP framework uses the Session Initiation
Protocol (SIP) [3] to set up, control and tear down the
connections between users.
Media is transported using the Real-Time Transport Protocol (RTP)
in the manner described in RFC4103.
This framework allows for implementation of services that meet the
requirement of providing a text-based conversational service,
equivalent to voice based telephony. In particular, ToIP offers an
IP equivalent of text telephony services as used by deaf, hard of
hearing and speech-impaired individuals.
In addition, real-time text conversations can be combined with
other conversational services using different media like video or
voice.
By using SIP, ToIP allows participants to negotiate all media
including real-time text conversation[4, 5]. This is a highly
desirable function for all IP telephony users, but essential for
deaf, hard of hearing, or speech impaired people who have limited
or no use of the audio path of the call.
It is important to understand that real-time text conversations
are significantly different from other text-based communications
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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 takes place, thus providing the same mode of
interaction as voice telephony does for hearing people. Store-and-
forward systems like email or messaging on mobile networks or non-
streaming systems like instant messaging are unable to provide
that functionality.
5.2. Requirements for ToIP
In order to make ToIP the equivalent of what voice is to hearing
people, it needs to offer equivalent features in terms of
conversationality as voice telephony provides to hearing people.
To achieve that, ToIP MUST:
a. Offer real-time presentation of the conversation;
b. Provide simultaneous transmission in both directions;
c. Provide interoperability with text conversation features in
other networks, for instance the PSTN, accepting functional
limitations that will occur during interoperation.
d. Not prevent other media, like audio and video, to be used in
conjunction with ToIP.
Users might want to use multiple modes of communication during the
conversation, either at the same time or by switching between
modes, e.g., between text and audio for example. Native ToIP
services MUST ensure that the text interface is always available.
When communicating via a gateway to other networks and protocols,
the service SHOULD support all the functionality for alternating
or simultaneous use of modalities as offered by the destination
network.
ToIP will often be used to access a relay service [I], allowing
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.
5.3. Use of SIP and RTP
ToIP services MUST use the Session Initiation Protocol (SIP) [3]
for setting up, controlling and terminating sessions for real-time
text conversation with one or more participants and possibly
including other media like video or audio.
Thus, participants are allowed to negotiate on a set of compatible
media types with session descriptions used in SIP invitations. A
ToIP service MUST always support at least one Text media type.
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ToIP services MUST use the Real-Time Transport Protocol (RTP)
according to the specification of RFC4103 for the transport of
text between participants, which implements T.140 on IP networks.
The standardized T.140 real-time text conversation [4], in
addition to audio and video communications, will be a valuable
service to many, including on non-IP networks. Real-time text can
be expressed as a part of the session description in SIP and is a
useful subset of Total Conversation.
The ToIP specification describes a 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.
If the User Agents of different participants indicate that there
is an incompatibility between their capabilities to support
certain media types, e.g. one terminal only offering T.140 over IP
as described in RFC4103 and the other one only supporting audio,
the user might want to invoke a transcoding services.
Examples of possible scenarios for including a relay service in
the conversation are: speech-to-text (STT), text-to-speech (TTS),
text bridging after conversion from speech, audio bridging after
conversion from text, etc.
The session description protocol (SDP) [6] used in SIP to describe
the session is used to express these attributes of the session
(e.g., uniqueness in media mapping for conversion from one media
to another for each communicating party).
Real-time text can also be presented in conjunction with other
media like video and audio, as for example in Total Conversation
services.
User Agents providing ToIP functionality SHOULD provide suitable
alerting, specifically offering visual and/or tactile alerting so
that deaf and hard of hearing users can use them.
The SIP abilities to set up text conversation sessions from any
location, as well as privacy and security provisions SHOULD be
implemented in ToIP services.
Where ToIP is used in conjunction with other media, exposure of
SIP functions through the User Interface MUST be available in
equivalent fashion for all supported media. In other words, where
certain SIP call control functions are available for the audio
media part of the session, these functions MUST also be supported
for the text media part of the same session.
Any ToIP implementation MUST also allow invocation and use of
relevant transcoding services where these are available. This can
be achieved through application of SIP techniques for different
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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, ToIP services SHOULD support
interworking with text telephony [10].
The general framework for ToIP 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. Registration of user preferences
d. Negotiation procedures for device capabilities
e. Discovery and invocation of transcoding/translation services
between the media in the call
f. Different session establishment models for transcoding /
translation services invocation: Third party call control and
conference bridge model
g. 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
h. Media bridging services for T.140 real-time text as described
in RFC4103, audio, and video for multipoint communications
i. Transparent session setup, modification, and teardown between
text conversation capable and voice/video capable devices
j. Support of text media transport using T.140 over RTP as laid
out in RFC 4103 [4]
k. Signaling of status information, call progress and the like in
a suitable manner, bearing in mind the user may have a hearing
impairment
l. T.140 real-time text presentation mixing with voice and video
m. T.140 real-time text conversation sessions using SIP, allowing
users to move from one place to another
n. User privacy and security for sessions setup, modification, and
teardown as well as for media transfer
o. Interoperability between T.140 conversations and analogue text
telephones
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p. Routing of emergency calls according to national or regional
policy to the same level of a voice call.
5.4. Requirements for ToIP Interworking
Analog text telephony is cumbersome because of incompatible
national implementations where interworking was never considered.
A large number of these implementations have been documented in
ITU-T V.18, which also defines modem detection sequences for the
different text terminals. The full modem capability exchange
between two wildly different terminals can take more than one
minute to complete if both terminals have a common text
modulation.
To resolve international analog textphone incompatibilities, text
telephone gateways MUST transcode incoming analog signals into
T.140 and vice versa. The modem capability exchange time is then
also reduced, since V.18 allows the sequence of protocol discovery
to be customized. Hence, the text telephone gateways will assume
the analog text telephone protocol used in the region the gateway
is located. For example, in the USA, Baudot might be tried as the
initial protocol. If negotiation for Baudot fails, the full modem
capability exchange will then take place. In contrast, in the UK,
ITU-T V.21 might be the first choice.
6. Detailed requirements for Text-over-IP
ToIP services MUST use SIP for call control and signaling.
A ToIP user may wish to call another ToIP user, or join a
conference call involving several users. He or she may, also, wish
to initiate or join a multimedia call, such as a Total
Conversation call.
There may 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 routing and addressing.
Similarly, there are requirements that need to be satisfied during
call set up when other media are preferred by a user. For
instance, some users may prefer to use audio while others want to
use text as their preferred modality. In this case, transcoding
services might be needed 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 some of these requirements in
detail.
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6.1. Pre-Call Requirements
The need to use ToIP as a medium of communications can be
expressed by users 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 with a SIP
server that is part of the ToIP service.
b. Server to support User Preferences: SIP servers that are part
of ToIP services MUST have the capability to act on users
preferences for ToIP to accept or reject the call, based on the
user preferences defined during the pre-call setup registration
time. For example, if the user is called by another party, and it
is determined that a transcoding server is needed, the call MUST
be re-directed or otherwise handled accordingly.
6.2 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 a possible or preferred medium for conversation using SIP in
the session setup.
6.2.1 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 ToIP. For instance, the
conversation might be established using audio and the user could
subsequently elect to switch to text, or add text as an additional
modality, during the conversation. Systems supporting ToIP 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.
If the user requests simultaneous use of text and audio, 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. 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 specially formatted addresses or other
mechanisms for invoking gateways.
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The following features MAY need to be implemented to facilitate
the session establishment using ToIP:
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).
6.2.2 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 transcoding service MUST NOT require user
registration in any specific SIP registrar, but MAY require
authorisation of the SIP registrar in the service.
6.2.3 Alerting and session progress presentation
User Agents supporting ToIP MUST have an alerting method (e.g.,
for incoming calls) 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 by the User AgentÆs User
Interface 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 ToIP
originates from a gateway, this MAY be signaled to the user.
During a conversation that includes ToIP, 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
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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 SHOULD use simple language as much as
possible so that as many users as possible can understand it. 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 or 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.
In summary, it SHOULD 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.
6.2.4 Call Negotiations
The Session Description Protocol (SDP) used in SIP [3] provides
the capabilities to indicate ToIP as a media in the call setup.
RFC 4103 [5] provides the RTP payload type text/t140 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, SIPÆs
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].
6.2.5 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 ToIP,
this information MUST be available in text, and MAY be provided in
other visual media.
6.2.5.1 Answering Machine
Systems for ToIP MAY support an auto-answer function, equivalent
to answering machines on telephony networks. If an answering
machine function is supported, it MUST support at least 160
characters for the greeting message. It MUST support incoming text
message storage of a minimum of 4096 characters, although systems
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MAY support much larger storage. It is RECOMMENDED that systems
support storage of at least 20 incoming messages of up to 16000
characters.
When the answering machine is activated, user alerting SHOULD
still take place. The user SHOULD be allowed to monitor the auto-
answer progress and where this is provided the user MUST be
allowed to intervene during any stage of the answering machine and
take control of the session.
6.2.6 Actions During Calls
Certain actions need to be performed for the ToIP conversation
during the call and these actions are described 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
characters].
f. To enable the use of international character sets the
transmission format for text conversation SHALL be UTF-8, in
accordance with ITU-T T.140.
g. If text is detected to be missing after transmission, there
SHALL be an indication in the text marking the loss. For 7 bit
terminals this loss MAY be marked as an apostrophe: Æ.
g. When used from a terminal designed for PSTN text telephony, or
in interworking with such a terminal, ToIP shall enable
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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 the USA and the UK).
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.
6.2.6.1 Text and other Media Handling Between ToIP User Agents
The following requirements are valid for media handling during
calls:
a. When used between User Agents designed for ToIP, it SHALL be
possible to send and receive text simultaneously.
b. When used between User Agents that support 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.
c. It SHOULD be possible to know during the call that ToIP is
available, even if it is not invoked at call setup (only voice
and/or video is used for example). To disable this, the user must
disable the use of ToIP. This is possible during registration at
the REGISTRAR.
6.2.6.2 Call Action with Native ToIP User Agents
a. It SHOULD be possible to answer a call with text capabilities
enabled.
b. It MAY be possible to use video simultaneously with the other
media in the call.
c. It MUST be possible to answer a call in voice or video without
text enabled, and add text later in the call.
d. It MUST be possible to disconnect the call.
e. It SHOULD be possible to invoke multi-party calls.
f. It MUST be possible to transfer the call.
6.2.7 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 text calls.
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6.2.8 File storage
Systems that support ToIP MAY save the text conversation to a
file. This SHOULD be done using a standard file format. For
example: UTF8 text file in XML format including record timestamp,
party and the text conversation.
6.3 Conference Call Requirements for ToIP User Agents
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.
It SHOULD be possible to use the text medium in conference calls,
in a similar way as the audio is handled and the video is
displayed. Text in conferences can be used both for letting
individual participants use the text medium (for example, for
sidebar discussions in text while listening to the main conference
audio), as well as for central support of the conference with real
time text interpretation of speech.
6.4 Transport via RTP
ToIP uses RTP as the default transport protocol for transmission
of real-time text via medium text/t140 as specified in RFC 4103
[5].
The redundancy method of RFC 4103 [5] SHOULD be used for making
text transmission reliable.
Text capability MUST 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 98/98/98
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 optimized.
However, if good reasons exist, other transport mechanisms MAY be
offered and used for the T.140 coded text, provided that proper
negotiation is introduced, and RFC 4103 [5] transport MUST be used
as both the default as well as the fallback transport.
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6.5 Character Set
a. ToIP services MUST use UTF-8 encoding as specified in ITU-T
T.140 [12].
b. ToIP SHOULD handle characters with editing effect such as new
line, erasure and alerting during session as specified in ITU-T
T.140.
6.6 Transcoding
Transcoding of text may need to take place in gateways between
ToIP and other forms of text conversation. For example to connect
to a PSTN text telephone.
6.7 Relay Services
The relay service acts as an intermediary between two or more
callers using different media or different media encoding schemes.
The basic text relay service allows a translation of speech to
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 want to remind readers that there
exist other relay services like, for example, speech to sign
language and vice versa using video.
It is RECOMMENDED that ToIP implementations make the invocation
and use of relay services as easy as possible. It MAY happen
automatically when the call is being set up based on any valid
indication or negotiation of supported or preferred media types. 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. ToIP implementations
SHOULD support this transcoding framework.
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, the sign language being used (in the
video stream), etc.
It SHOULD be possible to route the call to a preferred relay
service even if the user makes the call from another region or
network than usually used.
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6.8 Emergency services
Access to emergency services using ToIP SHOULD provide an
equivalent service to the one offered by other supported media,
like audio.
6.9 User Mobility
ToIP User Agents SHOULD use the same mechanisms as other SIP User
Agents to resolve mobility issues. It is RECOMMENDED to use a SIP-
address for the users, resolved by a SIP REGISTRAR, to enable
basic user mobility. Further mechanisms are defined for the 3G IP
multimedia systems.
6.10 Confidentiality and Security
User 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] and SRTP
[19]
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.
7. Interworking Requirements for ToIP
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, especially for the PSTN text telephony to ensure
full backward interoperability with ToIP.
7.1 ToIP 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.
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.
A. van Wijk [Page 17 of 28]
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7.2 ToIP 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 ToIP 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. A buffer of at least 2K
characters 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-ToIP gateways MUST allow alternating use of text and voice.
PSTN and ISDN to ToIP 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.
7.3 ToIP 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 ToIP at the other side.
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7.3.1 "No-gain"
The "No-gain" text telephone transporting technology uses
specially modified EFR [15] and EVR [16] speech vocoders in both
mobile terminals used to provide a text telephony call. It
provides full duplex operation and supports alternating voice and
text.( "VCO/HCO"). It is dedicated to the CDMA and TDMA mobile
technologies and the US Baudot type of text telephones.
7.3.2 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.
7.3.3 "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.
7.3.4 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 on the mobile network. Gateways or the interworking
function provides interoperability with PSTN textphones.
7.3.5 Common Text Gateway Functions
Text gateways MUST cover 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. van Wijk [Page 19 of 28]
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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 or region are
called eg. "911" in USA and "112" in Europe. Routing text calls to
emergency services MAY require the use of a transcoding service.
A text gateway MUST act as a SIP User Agent on the IP side.
7.4 ToIP and Cellular Wireless ToIP
ToIP MAY be supported over the cellular wireless packet switched
service. It interfaces to the Internet. For 3GPP 3G services, the
support is described to use ToIP in 3G TS 26.235 [20].
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.
7.5 Instant Messaging Support
Many people use Instant Messaging to communicate via the Internet
using text. Instant Messaging transfers blocks of text rather than
streaming as is used by ToIP. As such, it is not a replacement for
ToIP and in particular does not meet the needs for real time
conversations of deaf, hard of hearing and speech-impaired users
as defined in RFC 3351 [21]. It is unsuitable for communications
through a relay service [I]. The streaming character of ToIP
provides a better user experience and, when given the choice,
users often prefer ToIP.
However, since some users might only have Instant Messaging
available, text gateways MAY be developed to allow interworking
between Instant Messaging systems and ToIP solutions.
Because Instant Messaging is based on blocks of text, rather than
on a continuous stream of characters, such gateways need to
transform between these two formats. Text gateways for
interworking between Instant Messaging and ToIP MUST concatenate
individual characters originating at the ToIP side into blocks of
text and:
a. When the length of the concatenated message becomes longer than
50 characters, the buffered text SHOULD be transmitted to the
Instant Messaging side as soon as any non-alphanumerical character
is received from the ToIP side.
b. When a new line is received from the ToIP side, the buffered
characters up to that point, including the carriage return and/or
line feed characters, SHOULD be transmitted to the Instant
Messaging side.
A. van Wijk [Page 20 of 28]
draft-ietf-sipping-ToIP-01.txt July 18 2005
c. When the ToIP side has been idle for at least 5 seconds, all
buffered text up to that point SHOULD be transmitted to the
Instant Messaging side.
It is RECOMMENDED that during the session, both users are
constantly updated on the progress of the text input.
Many Instant Messaging protocols signal that a user is typing to
the other party in the conversation. Text gateways between such
Instant Messaging protocols and ToIP MUST provide this signaling
to the Instant Messaging side when characters start being
received, or at the beginning of the conversation.
At the ToIP side, an indicator of writing the Instant Message MUST
be present where the Instant Messaging protocol provides one. For
example, the real-time text user MAY see . . . waiting for
replying IM. . . And per 5 seconds that pass a . (dot) can be
shown.
Those solutions will reduce the difficulties between a streaming
versus blocked text.
Even though the text gateway can connect Instant Messaging and
ToIP, the best solution is to take advantage of the fact that the
user interfaces and the user communities for instant messaging and
ToIP telephony are extremely similar. After all, the character
input, the character display, Internet connectivity and SIP stack
are the same for Instant Messaging (SIMPLE) and ToIP.
Devices that implement Instant Messaging SHOULD implement ToIP as
described in this document.
7.6 IP Telephony with Traditional RJ-11 Interfaces
Analogue adapters using SIP based IP communication and RJ-11
connectors for connecting traditional PSTN devices (ATA box)
SHOULD enable connection of legacy PSTN text telephones [18].
These adapters SHOULD contain V.18 modem functionality, voice
handling functionality, and conversion functions to/from SIP based
ToIP with T.140 transported according to RFC 4103 [5], in a
similar way as it provides interoperability for voice calls. If a
call is set up and text/t140 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 RFC 4103
[5] on board and thus act as a text gateway. Sending textphone
signals over the voice channel is undesirable due to possible
filtering and compression and packet loss between the end-points.
This can result in dropping characters in the textphone
conversation or even not allowing the textphones to connect with
each other.
A. van Wijk [Page 21 of 28]
draft-ietf-sipping-ToIP-01.txt July 18 2005
7.7 Multi-functional gateways
In practice many interworking gateways 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 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.
7.8 ToIP interoperability with PSTN text telephones.
Gateways between the ToIP network and other networks MAY need to
transcode text streams. ToIP makes use of the ISO 10646 character
set. Most PSTN textphones use a 7-bit character set, or a
character set that is converted to a 7-bit character set by the
V.18 modem.
When transcoding between character sets and T.140 in gateways,
special consideration MUST be given to the national variants of
the 7 bit codes, with national characters mapping into different
codes in the ISO 10 646 code space. The national variant to be
used could be selectable by the user on a per call basis, or be
configured as a national default for the gateway.
The missing text indicator in T.140, specified in T.140 amendment
1, cannot be represented in the 7 bit character codes. Therefore
these characters SHOULD be transcoded to the ' (apostrophe)
character in legacy text telephone systems, where this character
exists. For legacy systems where the character ' does not exist,
the . ( full stop ) character SHOULD be used instead.
7.9 Gateway Discovery
ToIP requires a method to invoke a text gateway. As described
previously in this draft, these text gateways MUST act as User
Agents at the IP side. The capabilities of the text gateway during
the call will be determined by the call capabilities of the
terminal that is using the gateway. For example, a PSTN textphone
is only able to receive voice and streaming text, so the text
gateway will only allow ToIP and audio.
Examples of possible scenarios for discovery of the text gateway
are:
A. van Wijk [Page 22 of 28]
draft-ietf-sipping-ToIP-01.txt July 18 2005
- PSTN textphone users dial a prefix number before dialing out.
- Separate text subscriptions, linked to the phone number or
terminal identifier/ IP address.
- Text capability indicators.
- Text preference indicator.
- Listen for V.18 modem modulation text activity in all calls.
- Call transfer request by the called user.
- Placing a call via the web, and using one of the methods
described here
- Text gateways with its own telephone number and/or SIP address.
(This requires user interaction with the text gateway to place a
call).
- ENUM address analysis and number plan
- Number or address analysis leads to the gateway for all PSTN
calls.
8. Afterword
The authors want to make it clear that ToIP is a way of allowing
real-time, interactive text conversation between all users and is
thus not only for the hearing and speech impaired users.
The users may invoke the ToIP services for many different reasons.
For example:
- 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.
- Hearing loss, tinnitus or deafness due to the aging process or
any other reason.
NOTE: In many of the above examples, text may accompany speech and
could be displayed in a manner similar to subtitling in
broadcasting environments or any other suitable manner. This
could occur for individuals who are hard of hearing and also for
mixed calls with a hearing and deaf person listening to the call.
9. Security Considerations
There are no additional security requirements other than described
earlier.
A. van Wijk [Page 23 of 28]
draft-ietf-sipping-ToIP-01.txt July 18 2005
10. Authors Addresses
The following people provided substantial technical and writing
contributions to this document, listed alphabetically:
Willem P. Dijkstra
TNO Informatie- en Communicatietechnologie
Postbus 15000
9700 CD Groningen
The Netherlands
Tel: +31 50 585 77 24
Fax: +31 50 585 77 57
Email: willem.dijkstra@tno.nl
Barry Dingle
ACIF, 32 Walker Street
North Sydney, NSW 2060 Australia
Tel +61 (0)2 9959 9111
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
Department of New Technologies
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
Henry Sinnreich
pulver.com
115 Broadhollow Rd
Suite 225
Melville, NY 11747
USA
Tel: +1.631.961.8950
A. van Wijk [Page 24 of 28]
draft-ietf-sipping-ToIP-01.txt July 18 2005
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
11. References
11.1 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. Hellstrom, "RTP Payload for Text Conversation, RFC 4103,
June 2005.
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 June 2005 - Work in progress.
8. G. Camarillo, H. Schulzrinne, E. Burger, and A. van Wijk,
"Transcoding Services Invocation in the Session Initiation
Protocol (SIP) Using Third Party Call Control (3pcc)" RFC 4117,
June 2005.
A. van Wijk [Page 25 of 28]
draft-ietf-sipping-ToIP-01.txt July 18 2005
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).
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.
19. Baugher, McGrew, Carrara, Naslund, Norrman, "The Secure Real
Time Transport Protocol (SRTP)", RFC 3711, IETF, March 2004.
20. IP Multimedia default codecs. 3GPP TS 26.235
21. Charlton, Gasson, Gybels, Spanner, van Wijk, "User
Requirements for the Session Initiation Protocol (SIP) in Support
of Deaf, Hard of Hearing and Speech-impaired Individuals", RFC
3351, IETF, August 2002.
22. J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with the
Session Description Protocol (SDP)", RFC 3624, IETF, June 2002.
A. van Wijk [Page 26 of 28]
draft-ietf-sipping-ToIP-01.txt July 18 2005
11.2 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.
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A. van Wijk [Page 27 of 28]
draft-ietf-sipping-ToIP-01.txt July 18 2005
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