SIPPING Workgroup A. van Wijk (editor)
Internet-Draft Viataal
Category: Informational
Expires: March 6 2006 September 7 2005
Framework of requirements for real-time text conversation using SIP
draft-ietf-sipping-toip-03.txt
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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 Real-Time
Transport Protocol. It discusses requirements for real-time Text-
over-IP as well as interworking between Text-over-IP and existing
text telephony on the PSTN and other networks.
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Table of Contents
1. Introduction.....................................................3
2. Scope............................................................4
3. Terminology......................................................4
4. Definitions......................................................4
5. Framework Description............................................6
5.1. General requirements for ToIP..................................6
5.1.1 General ToIP Summary..........................................8
5.2. General Requirements for ToIP Interworking.....................8
5.2.1 PSTN Interworking.............................................9
5.2.2 Cellular circuit switched Text-Telephony.....................10
5.2.2.1 Cellular "No-gain".........................................10
5.2.2.2 Cellular Text Telephone Modem (CTM)........................10
5.2.2.3 Cellular "Baudot mode".....................................11
5.2.3 Cellular data channel mode...................................11
5.2.4 Cellular Wireless ToIP.......................................11
5.2.5 Instant Messaging Support....................................11
6. Detailed requirements for ToIP..................................11
6.1. Pre-Session Requirements......................................12
6.2 Basic Point-to-Point Session Requirements......................12
6.2.1 Session control..............................................12
6.2.2 Text transport...............................................12
6.2.3 Session Setup................................................13
6.2.4 Addressing...................................................13
6.2.5 Alerting.....................................................14
6.2.6 Session information..........................................14
6.2.7 Session progress information.................................14
6.2.8 Session Negotiations.........................................15
6.2.9 Answering....................................................15
6.2.9.1 Answering Machine..........................................15
6.2.10 Actions During a Session....................................15
6.2.10.1 Text Transport............................................16
6.2.10.2 Handling Text and other Media.............................16
6.2.11 Additional session control..................................17
6.2.12 File storage................................................17
6.3 Conference Session Requirements................................17
6.4 Real-time Editing and User Alerting............................17
6.5 Emergency services.............................................17
6.6 User Mobility..................................................18
6.7 Firewalls and NATs.............................................18
7. Interworking Requirements for ToIP..............................18
7.1 ToIP Interworking Gateway Services.............................18
7.2 ToIP and PSTN/ISDN Text-Telephony Interworking.................18
7.3 ToIP and Cellular Wireless ToIP................................19
7.4 Instant Messaging Support......................................19
7.5 Common Text Gateway Functions..................................20
7.5.1 Protocol support.............................................20
7.5.2 Relay buffer storage.........................................20
7.5.3 Emergency calls through gateways.............................21
7.5.4 Text Gateway Invocation......................................21
7.6 Home Gateways or Analog Terminal Adapters......................21
7.7 Multi-functional Combination gateways..........................22
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7.8 Transcoding....................................................22
7.9 Relay Services.................................................23
7.9.1 Basic function of the relay service..........................23
7.9.2 Invocation of relay services.................................23
8. Security Considerations.........................................23
9. Authors Addresses...............................................24
10. References.....................................................25
10.1 Normative references..........................................25
10.2 Informative references........................................27
1. Introduction
For many years, text has been in use as a medium for
conversational, interactive dialogue between users in a similar
way to how voice telephony is used. Such interactive text is
different from messaging and semi-interactive solutions like
Instant Messaging in that it offers an equivalent conversational
experience to users who cannot, or do not wish to, use voice. It
therefore meets a different set of requirements from other text-
based solutions already available on IP networks.
Traditionally, deaf, hard of hearing and speech-impaired people
are amongst the most prolific users of conversational, interactive
text but, because of its interactivity, it is becoming popular
amongst mainstream users as well.
This document describes how existing IETF protocols can be used to
implement a Text-over-IP solution (ToIP). This ToIP framework is
specifically designed to be compatible with Voice-over-IP (VoIP)
environments, as well as meeting the userÆs requirements,
including those of deaf, hard of hearing and speech-impaired users
as described in RFC3351 [19].
The Session Initiation Protocol (SIP) is the protocol of choice
for control of Multimedia communications and Voice-over-IP (VoIP)
in particular. It offers all the necessary control and signaling
required for the ToIP framework.
The Real-Time Transport Protocol (RTP) 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, multi-media services, including
Total Conversation.
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2. Scope
This document defines a framework for the implementation of real-
time ToIP, either stand-alone or as a part of multimedia services,
including Total Conversation. It defines the:
a. Requirements of Real-time, interactive text;
b. Requirements for ToIP interworking;
c. Description of ToIP using SIP and RTP;
d. Description of ToIP interworking with other text services.
3. Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT
RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as
described in BCP 14, RFC 2119 [2] and indicate requirement levels
for compliant implementations.
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.
Cellular - Telephone systems based on radio transmission to become
wireless. Also called Wireless or Mobile systems.
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.
Textphone û also "text telephone". A terminal device that allows
end-to-end real-time, interactive text communication using analog
transmission. A variety of PSTN textphone protocols exists world-
wide. A textphone can often be combined with a voice telephone, or
include voice communication functions for simultaneous or
alternating use of text and voice in a call.
Text bridging - a function of a gateway service that enables the
flow of text through the service between the users involved in the
call.
Text gateway - a function that transcodes between different forms
of text transport methods, e.g., between ToIP in IP networks and
Baudot or ITU-T V.21 text telephony in the PSTN.
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Text Relay Service - a third-party or intermediary that enables
communications between deaf, hard of hearing and speech-impaired
people, and voice telephone users by translating between voice and
text in a call.
Text telephony û analog textphone service.
Total Conversation - a multimedia service offering real time
conversation in video, text and voice according to interoperable
standards. All media flow in real time. (See ITU-T F.703
"Multimedia conversational services".)
Transcoding Services - services of a third-party user agent that
transcodes one stream into another. Transcoding can be done by
human operators, in an automated manner or a combination of both
methods. Text Relay Services are examples of a transcoding service
between text and audio.
TTY û alternative designation for a text telephone or textphone,
often used in USA. Also called TDD, Telecommunication Device for
the Deaf.
Video Relay Service - A service that enables communications
between deaf and hard of hearing people, and hearing persons with
voice telephones by translating between sign language and spoken
language in a call.
Acronyms:
2G Second generation cellular (mobile)
2.5G Enhanced second generation cellular (mobile)
3G Third generation cellular (mobile)
CDMA Code Division Multiple Access
CLI Calling Line Identification
CTM Cellular Text Telephone Modem
ENUM E.164 number storage in DNS (see RFC3761)
GSM Global System of Mobile Communication
ISDN Integrated Services Digital Network
ITU-T International Telecommunications Union-Telecommunications
Standardisation Sector
NAT Network Address Translation
PSTN Public Switched Telephone Network
RTP Real Time Transport Protocol
SDP Session Description Protocol
SIP Session Initiation Protocol
SRTP Secure Real Time Transport Protocol
TDD Telecommunication Device for the Deaf
TDMA Time Division Multiple Access
TTY Analog textphone (Teletypewriter)
ToIP Text over Internet Protocol
UTF-8 Universal Transfer Format-8
VCO/HCO Voice Carry Over/Hearing Carry Over
VoIP Voice over Internet Protocol
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5. Framework Description
This framework defines the requirements of a text-based
conversational service that is the text equivalent of voice based
telephony. Real-time text conversation can be combined with other
conversational services like video or voice.
ToIP also offers an IP equivalent of analog text telephony
services as used by deaf, hard of hearing and speech-impaired
individuals.
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 takes place. This provides 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. In particular, they do not
allow for smooth communication through a Text Relay Service.
This framework uses existing standards that are already commonly
used for voice based conversational services on IP networks. It
uses the Session Initiation Protocol (SIP) to set up, control and
tear down the connections between users whilst the media is
transported using the Real-Time Transport Protocol (RTP) as
described in RFC4103 [5].
This framework is designed to meet the requirements of RFC3351
[19]. As such, it offers a standardized way for offering text-
based, conversational services that can be used as an equivalent
to voice telephony by deaf, hard of hearing and speech-impaired
individuals.
SIP 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.
5.1. General requirements for ToIP
In order to make ToIP the text equivalent of voice services, it
needs to offer equivalent features in terms of conversationality
as voice telephony provides. To achieve that, ToIP needs to:
a. Offer real-time presentation of the conversation;
b. Provide simultaneous transmission in both directions;
c. Support both point-to-point and multipoint communication;
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d. Allow other media, like audio and video, to be used in
conjunction with ToIP;
e. Ensure that the text service is always available.
Real-time text is a useful subset of Total Conversation defined in
ITU-T F.703 [23]. Users could use multiple modes of communication
during the conversation, either at the same time or by switching
between modes, e.g., between text and audio.
Users may invoke ToIP services for many different reasons:
- Because they are in a noisy environment, e.g., in a machine room
of a factory where listening is difficult.
- Because they are busy with another call and want to participate
in two calls at the same time.
- For implementing text and/or speech recording services (e.g.,
text documentation/ audio recording for
legal/clarity/flexibility purposes).
- To overcome language barriers through speech translation and/or
transcoding services.
- Because of hearing loss, deafness or tinnitus as a result of the
aging process or for any other reason, thus creating a need to
replace or complement voice with text in conversational
sessions.
NOTE: In many of the above examples, text may accompany speech.
The text could be displayed side by side, in a manner similar to
subtitling in broadcasting environments, or in any other suitable
manner. This could occur for users who are hard of hearing and
also for mixed media calls with both hearing and deaf people
participating in the call.
User Agents providing ToIP functionality need to provide suitable
alerting indications, specifically offering visual and/or tactile
alerting for deaf and hard of hearing users.
The ability of SIP to set up conversation sessions from any
location, as well as its privacy and security provisions, MUST be
maintained by ToIP services.
Where ToIP is used in conjunction with other media, exposure of
SIP functions through the User Interface needs to be done in an
equivalent manner 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. For example, call
transfer must act on all media in the session.
T.140 real-time text conversation [4], in addition to audio and
video communications, is a valuable service for many users,
including those on non-IP networks. T.140 also provides for real-
time editing of the text.
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5.1.1 General ToIP Summary
The general requirements for ToIP are:
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. Support of text media transport using T.140 over RTP as
described in RFC 4103 [5]
f. Signaling of status information, call progress and the like in
a suitable manner, bearing in mind that the user may have a
hearing impairment
g. T.140 real-time text presentation mixing with voice and video
h. T.140 real-time text conversation sessions using SIP, allowing
users to move from one place to another
i. User privacy and security for sessions setup, modification, and
teardown as well as for media transfer
j. Routing of emergency calls according to national or regional
policy with the same level of functionality as a voice call.
5.2. General Requirements for ToIP Interworking
This section describes the general ToIP interworking requirements
and gives some background information to many of the issues.
There is a range of existing text services. There is also a range
of network technologies that could support text services (see
examples below). ToIP needs to provide interoperability with text
conversation features in other networks, for instance the PSTN,
and with some text messaging services.
Text gateways are used for converting between different media
types. They could be used between networks or within networks
where different transport technologies are used.
When communicating via a gateway to other networks and protocols,
the ToIP service SHOULD support the functionality for alternating
or simultaneous use of modalities as offered by the destination
network.
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Address information, both called and calling, SHOULD be
transferred, and possibly converted, when interworking between
different networks.
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 minimal. In particular,
buffering of whole lines of text will not meet character delay
requirements.
If the User Agents of different participants indicate that there
is an incompatibility between their capabilities to support
certain media types, e.g. one terminal only offering T.140 over IP
as described in RFC4103 [5] and the other one only supporting
audio, the user might want to invoke a transcoding service.
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 general requirements for ToIP Interworking are:
a. Interoperability between T.140 conversations [4] and analog
text telephones
b. Discovery and invocation of transcoding/translation services
between the media in the call
c. Different session establishment models for transcoding /
translation services invocation: Third party call control and
conference bridge model
d. 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
e. Media bridging services for T.140 real-time text, as described
in RFC4103 [5], audio and video for multipoint communications
f. Transparent session setup, modification, and teardown between
text conversation capable devices and voice/video capable
devices
g. Buffering of text when interworking with media that transport
text at different rates.
5.2.1 PSTN Interworking
Analog text telephony is cumbersome because of incompatible
national implementations where interworking was never considered.
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A large number of these implementations have been documented in
ITU-T V.18 [10], which also defines the modem detection sequences
for the different text protocols. The modem type identification
may in rare cases take considerable time depending on user
actions.
To resolve analog textphone incompatibilities, text telephone
gateways are needed to transcode incoming analog signals into
T.140 and vice versa. The modem capability exchange time can be
reduced by the text telephone gateways initially assuming the
analog text telephone protocol used in the region where the
gateway is located. For example, in the USA, Baudot [III] might be
tried as the initial protocol. If negotiation for Baudot fails,
the full V.18 modem capability exchange will take place. In the
UK, ITU-T V.21 [II] might be the first choice.
5.2.2 Cellular 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.
5.2.2.1 Cellular "No-gain"
The "No-gain" text telephone transporting technology uses
specially modified EFR [13] and EVR [14] speech vocoders in mobile
terminals used to provide a text telephony call. It provides full
duplex operation and supports alternating voice and text
("VCO/HCO"). It is dedicated to CDMA and TDMA mobile technologies
and the US Baudot (i.e. 45 bit/s) type of text telephones.
5.2.2.2 Cellular Text Telephone Modem (CTM)
CTM [15] is a technology independent modem technology that
provides the transport of text telephone characters at up to 10
characters/sec using modem signals that can be carried by many
voice codecs and uses a highly redundant encoding technique to
overcome the fading and cell changing losses.
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5.2.2.3 Cellular "Baudot mode"
This term is often used by cellular terminal suppliers for a GSM
cellular phone mode that allows TTYs to operate into a cellular
phone and to communicate with a fixed line TTY.
5.2.3 Cellular 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 provide interoperability
with PSTN textphones.
5.2.4 Cellular Wireless ToIP
ToIP could be supported over cellular wireless packet switched
services that interface to the Internet. For 3GPP 3G services, the
support is described to use ToIP in 3G TS 26.235 [18]. Low data
rates and additional delays can affect performance.
5.2.5 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 including those of deaf, hard of hearing and speech-
impaired users as defined in RFC 3351 [19]. It is unsuitable for
communications through a relay service [I]. The streaming nature
of ToIP provides a more direct conversational user experience and,
when given the choice, users may prefer ToIP.
Text gateways could be developed to allow interworking between
Instant Messaging systems and ToIP solutions.
6. Detailed requirements for ToIP
A ToIP user may wish to call another ToIP user, or join a
conference session involving several users or initiate or join a
multimedia session, such as a Total Conversation session.
There may be some need for pre-session setup e.g. storing of
registration information in the SIP registrar, to provide
information about how a user can be contacted. This will allow
sessions to be set up rapidly and with proper routing and
addressing.
Similarly, there are requirements that need to be satisfied during
session set up when other media are preferred by a user. For
instance, some users may indicate their preferred modality to be
audio while others may indicate text. In this case, transcoding
services might be needed for text-to-speech (TTS) and speech-to-
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text (STT). The requirements for transcoding services need to be
negotiated in real-time to set up the session.
The subsequent subsections describe some of these requirements in
detail.
6.1. Pre-Session Requirements
The need to use text as a medium of communications can be
expressed by users during registration time. Two situations need
to be considered in the pre-session setup environment:
a. User Preferences: It MUST be possible for a user to indicate a
preference for text by registering that preference with a SIP
server that is part of the ToIP service.
b. Server to support User Preferences: SIP servers that support
ToIP services MUST have the capability to act on calling user
preferences for text in order to accept or reject the session-,
based on the called userÆs preferences defined as part of the
pre-session setup registration. For example, if the user is
called by another party, and it is determined that a
transcoding server is needed, the session MUST be re-directed
or otherwise handled accordingly.
6.2 Basic Point-to-Point Session Requirements
A point-to-point session takes place between two parties. The
requirements are described in subsequent sub-sections. They assume
that one or both of the communicating parties will indicate text
as a possible or preferred medium for conversation using SIP in
the session setup.
6.2.1 Session control
ToIP services MUST use the Session Initiation Protocol (SIP) [3]
for setting up, controlling and terminating sessions for real-time
text conversation with one or more participants and possibly
including other media like video or audio. The session description
protocol (SDP) [6] used in SIP to describe the session is used to
express the attributes of the session and to negotiate a set of
compatible media types.
6.2.2 Text transport
A ToIP service MUST always support at least one Text media type.
ToIP services MUST support the Real-Time Transport Protocol (RTP)
[24] according to the specification of RFC4103 [5] for the
transport of text between participants.
RFC4103 describes the transmission of T.140 [4] on IP networks.
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6.2.3 Session Setup
Users will set up a session by identifying the remote party or the
service they want to connect to. However, conversations could be
started using a mode other than text. 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 constraints in the network, the system
MUST try to establish communication with best effort. If the user
has expressed a preference for text, establishment of a connection
including text MUST have priority over other outcomes of the
session setup.
The following features MAY be implemented to facilitate the
session establishment using ToIP:
a. Caller Preferences: SIP headers (e.g., Contact)[24] 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 session should be
handled either using text as a preferred medium or not, and
whether a designated SIP proxy needs to handle this session or
it will be handled in the SIP user agent.
c. SIP Server support for User Preferences: SIP servers can also
handle the incoming sessions in accordance with preferences
expressed for ToIP. The SIP Server can also enforce ToIP policy
rules for communications (e.g. use of the transcoding server
for ToIP).
6.2.4 Addressing
The SIP [3] addressing schemes MUST be used for all entities in a
ToIP session. For example, SIP URLÆs or Tel URLÆs are 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 to invoke the service.
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6.2.5 Alerting
User Agents supporting ToIP MUST have an alerting method (e.g.,
for incoming sessions) that can be used by deaf and hard of
hearing people or provide a range of alternative, but equivalent,
alerting methods that can be selected by all users, regardless of
their abilities.
It should be noted that external alerting systems exist and one
common interface for triggering the alerting action is a contact
closure between two conductors.
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.
6.2.6 Session information
If present, identification of the originating party (for example
in the form of a URL or a CLI) MUST be clearly presented to the
user in a form suitable for the user BEFORE the session invitation
is answered. When a session invitation involving ToIP originates
from a gateway, this MAY be signaled to the user.
The user MUST be informed of any change in modalities.
6.2.7 Session progress information
During a conversation that includes ToIP, status and session
progress information MUST be provided in a textual form so users
can perform all session control functions. That information MUST
be equivalent to session progress information delivered in any
other format, for example audio.
Session progress information SHOULD use simple language so that as
many users as possible can understand it. The use of jargon or
ambiguous terminology SHOULD be avoided. It is RECOMMENDED that
text information be used together with icons to symbolise the
session progress information.
There MUST be a clear indication, in a modality useful to the
user, whenever a session is connected or disconnected. A user
SHOULD never be in doubt about the status of the session, even if
the user is unable to make use of the audio or visual indication.
For example, tactile indications could be used by deafblind
individuals.
In summary, it SHOULD be possible to observe indicators about:
- Incoming session
- Availability of text, voice and video channels
- Session progress
- Incoming text
- Any loss in incoming text
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- Typed and transmitted text.
For users who cannot use the audible alerter for incoming
sessions, it is RECOMMENDED to include a tactile as well as a
visual indicator.
6.2.8 Session Negotiations
The Session Description Protocol (SDP) used in SIP [3] provides
the capabilities to indicate text as a medium in the session
setup. RFC 4103 [5] uses the RTP payload type "text/t140" for
support of ToIP which can be indicated in the SDP as a part of the
SIP INVITE, OK and SIP/200/ACK media negotiations. In addition,
SIPÆs offer/answer model [20] can also be used in conjunction with
other capabilities including the use of a transcoding server for
enhanced session negotiations [7,8,9].
6.2.9 Answering
Systems SHOULD provide a best-effort approach to answering
invitations for session set-up and users SHOULD be informed when
the session is accepted by the other party. On all systems that
both inform users of session status and support ToIP, this
information MUST be available in textual form and MAY also be
provided in other media.
6.2.9.1 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
MAY support much larger storage. It is RECOMMENDED that systems
support storage of at least 20 incoming messages of up to 16000
characters per message.
When the answering machine is activated, user alerting SHOULD
still take place. The user SHOULD be allowed to monitor the auto-
answer progress and where this is provided the user SHOULD be
allowed to intervene during any stage of the answering machine
procedure and take control of the session.
6.2.10 Actions During a Session
Certain actions need to be performed during ToIP conversation:
a. Text transmission from a terminal SHALL be performed character
by character as entered, or in small groups of characters, so
that no character is delayed from entry to transmission by more
than 300 milliseconds.
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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. To enable the use of international character sets, the
transmission format for text conversation SHALL be UTF-8 [12],
in accordance with ITU-T T.140.
d. If text is detected to be missing after transmission, there
SHOULD be a "text loss" indication in the text as specified in
T.140 Addendum 1 [4].
e. When the display of text conversation is included in the design
of the end user equipment, the display of the dialogue SHOULD
be made so that it is easy to differentiate the text belonging
to each party in the conversation.
6.2.10.1 Text Transport
ToIP uses RTP as the default transport protocol for the
transmission of real-time text via the medium "text/t140" as
specified in RFC 4103 [5].
The redundancy method of RFC 4103 [5] SHOULD be used to
significantly increase the reliability of the text transmission. A
redundancy level using 2 generations gives very reliable results
and is therefore RECOMMENDED.
Text capability MUST be announced in SDP by a declaration similar
to 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
By having this single coding and transmission scheme for real time
text defined in the SIP session control environment, the
opportunity for interoperability is optimized. However, if good
reasons exist, other transport mechanisms MAY be offered and used
for the T.140 coded text provided that proper negotiation is
introduced, but RFC 4103 [5] transport MUST be used as both the
default and the fallback transport.
6.2.10.2 Handling Text and other Media.
A call is one or more related sessions. The following requirements
apply to media handling during a call:
a. When used between User Agents designed for ToIP, it SHALL be
possible to send and receive text simultaneously.
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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 a call that ToIP is
available, even if it is not invoked at call setup (e.g. when
only voice and/or video is used initially). To disable this,
the user MUST disable the use of ToIP. This is possible during
registration at the SIP registrar.
6.2.11 Additional session control
Systems that support additional session control features, for
example call waiting, forwarding, hold etc on voice sessions, MUST
offer this functionality for text sessions.
6.2.12 File storage
Systems that support ToIP MAY save the text conversation to a
file. This SHOULD be done using a standard file format. For
example: a UTF8 text file in XML format [11] including timestamps,
party names (or addresses) and the text conversation.
6.3 Conference Session Requirements
The conference session requirements deal with multipoint
conferencing sessions where there will be one or more ToIP capable
devices and/or other end user devices where the total number of
end user devices will be at least three.
It SHOULD be possible to use the text medium in conference
sessions in a similar way to how audio is handled and 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 Real-time Editing and User Alerting
ToIP SHOULD handle characters such as new line, erasure and
alerting during a session as specified in ITU-T T.140.
6.5 Emergency services
It MUST be possible to place an emergency call using ToIP and it
MUST be possible to use a relay service in such call. The
emergency service provided to users utilising the text medium MUST
be equivalent to the emergency service provided to users utilising
speech or other media.
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6.6 User Mobility
ToIP User Agents SHOULD use the same mechanisms as other SIP User
Agents to resolve mobility issues. It is RECOMMENDED that users
use a SIP-address, resolved by a SIP registrar, to enable basic
user mobility. Further mechanisms are defined for all session
types for 3G IP multimedia systems.
6.7 Firewalls and NATs
ToIP uses the same signaling and transport protocols as VoIP
hence, the same firewall and NAT solutions and network
functionality that apply to VoIP MUST also apply to ToIP.
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 the interoperability
requirements, especially for 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
(due to a hearing impairment), or who can hear but must send text
(due to a speech impairment).
Session setup through gateways to other networks MAY require the
use of specially formatted addresses or other mechanisms for
invoking those gateways.
Different data rates of different protocols MAY require text
buffering.
Transcoding of text to and from other coding formats MAY need to
take place in gateways between ToIP and other forms of text
conversation, for example to connect to a PSTN text telephone.
7.2 ToIP and PSTN/ISDN Text-Telephony Interworking
On PSTN networks, transmission of interactive text takes place
using a variety of codings and modulations, including ITU-T V.21
[II], Baudot [III], DTMF, V.23 [IV] and others. Many difficulties
have arisen as a result of this variety in text telephony
protocols and the ITU-T V.18 [10] standard was developed to
address some of these issues.
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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.
V.18 MUST be supported on the PSTN side of a PSTN-ToIP gateway.
PSTN-ToIP gateways MUST allow alternating use of text and voice if
the PSTN textphone involved at the PSTN side of the session
supports this. (This mode is often called VCO/HCO).
Calling party identification information, such as CLI, MUST be
passed by gateways and converted to an approapriate form if
required.
7.3 ToIP and Cellular Wireless ToIP
ToIP MAY be supported over the cellular wireless packet switched
service. It interfaces to the Internet.
A text gateway with cellular wireless packet switched services
MUST be able to route text calls to emergency service providers
when any of the recognized emergency numbers that support text
communication for the country.
7.4 Instant Messaging Support
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, gateways MUST transcode between the 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 indicator is received from the ToIP side, the
buffered characters up to that point, including the carriage
return and/or line feed characters, SHOULD be transmitted to
the Instant Messaging side.
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.
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Many Instant Messaging protocols signal that a user is typing to
the other party in the conversation. Text gateways between such
Instant Messaging protocols and ToIP MUST provide this signaling
to the Instant Messaging side when characters start being
received, or at the beginning of the conversation.
At the ToIP side, an indicator of writing the Instant Message MUST
be present where the Instant Messaging protocol provides one. For
example, the real-time text user MAY see ". . . waiting for
replying IM. . . " and when 5 seconds have passed another . (dot)
can be shown.
Those solutions will reduce the difficulties between streaming and
blocked text services.
Even though the text gateway can connect Instant Messaging and
ToIP, the best solution is to take advantage of the fact that the
user interfaces and the user communities for instant messaging and
ToIP telephony are very similar. After all, the character input,
the character display, Internet connectivity and SIP stack are the
same for Instant Messaging (SIMPLE) and ToIP.
Devices that implement Instant Messaging SHOULD implement ToIP as
described in this document so that a more complete text
communication service can be provided.
7.5 Common Text Gateway Functions
Text gateways MUST allow for the differences that result from
different text protocols. The protocols to be supported will
depend on the service requirements of the Gateway.
7.5.1 Protocol support
Text gateways MUST use the ITU-T V.18 [10] standard at the PSTN
side. A text gateway MUST act as a SIP User Agent on the IP side
and support RFC4103 text transport.
7.5.2 Relay buffer storage
When text gateway functions are invoked, there will be a need for
intermediate storage of characters before transmission to a device
receiving text slower than the transmitting speed of the sender.
Such temporary storage SHALL be dimensioned to adjust for
receiving at 30 characters per second and transmitting at 6
characters per second for up to 4 minutes (i.e. less than 3k
characters).
Interoperation of half-duplex and full-duplex protocols MAY
require text buffering. Some intelligence will be needed to
determine when to change direction when operating in half-duplex
mode. Identification may be required of half-duplex operation
either at the "user" level (ie. users must inform each other) or
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at the "protocol" level (where an indication must be sent back to
the Gateway).
7.5.3 Emergency calls through gateways
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 e.g. "911" in USA and "112" in Europe. Routing text calls
to emergency services MAY require the use of a transcoding
service.
7.5.4 Text Gateway Invocation
ToIP interworking 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 generally only able to receive voice
and streaming text, so the text gateway will only allow ToIP and
audio.
Examples of possible scenarios for invocation of the text gateway
are:
a. PSTN textphone users dial a prefix number before dialing out.
b. Separate text subscriptions, linked to the phone number or
terminal identifier/ IP address.
c. Text capability indicators.
d. Text preference indicator.
e. Listen for V.18 modem modulation text activity in all PSTN
calls and routing of the call to an appropriate gateway.
f. Call transfer request by the called user.
g. Placing a call via the web, and using one of the methods
described here
h. Text gateways with its own telephone number and/or SIP address.
(This requires user interaction with the text gateway to place
a call).
i. ENUM address analysis and number plan
j. Number or address analysis leads to a gateway for all PSTN
calls.
7.6 Home Gateways or Analog Terminal Adapters
Analog terminal adapters (ATAs) using SIP based IP communication
and RJ-11 connectors for connecting traditional PSTN devices
SHOULD enable connection of legacy PSTN text telephones [16].
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 sessions.
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If a session is set up and text/t140 capability is not declared by
the destination endpoint (by the end-point terminal or the text
gateway in the network at the end-point), a method for invoking a
transcoding server SHALL be used. If no such server is available,
the signals from the textphone MAY be transmitted in the voice
channel as audio with high quality of service.
NOTE: It is preferred that such analog terminal adaptors do use
RFC 4103 [5] on board and thus act as a text gateway. Sending
textphone signals over the voice channel is undesirable due to
possible filtering and compression and packet loss between the
end-points. This can result in character loss in the textphone
conversation or even not allowing the textphones to connect to
each other.
7.7 Multi-functional Combination gateways
In practice many interworking gateways will be implemented as
gateways that combine different functions. As such, a text gateway
could be built to have modems to interwork with the PSTN and
support both Instant Messaging as well as ToIP. Such interworking
functions are called Combination gateways.
Combination gateways MUST provide interworking between all of
their supported text based functions. For example, a text gateway
that has modems to interwork with the PSTN and that support both
Instant Messaging and 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 Transcoding
Gateways between the ToIP network and other networks MAY need to
transcode text streams. ToIP makes use of the ISO 10646 character
set. Most PSTN textphones use a 7-bit character set, or a
character set that is converted to a 7-bit character set by the
V.18 modem.
When transcoding between character sets and T.140 in gateways,
special consideration MUST be given to the national variants of
the 7 bit codes, with national characters mapping into different
codes in the ISO 10646 code space. The national variant to be used
could be selectable by the user on a per call basis, or be
configured as a national default for the gateway.
The indicator of missing text in T.140, specified in T.140
amendment 1, cannot be represented in the 7 bit character codes.
Therefore the indicator of missing text SHOULD be transcoded to
the ' (apostrophe) character in legacy text telephone systems,
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where this character exists. For legacy systems where the
character ' does not exist, the . ( full stop ) character SHOULD
be used instead.
7.9 Relay Services
The relay service acts as an intermediary between two or more
callers using different media or different media encoding schemes.
7.9.1 Basic function of the relay service
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 other
relay services exist, like video relay services transcoding speech
to sign language and vice versa where the signing is communicated
using video.
7.9.2 Invocation of relay services
It is RECOMMENDED that ToIP implementations make the invocation
and use of relay services as easy as possible. It MAY happen
automatically when the session is being set up based on any valid
indication or negotiation of supported or preferred media types. A
transcoding framework document using SIP [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 session.
When setting up a session, 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 session to a preferred relay
service even if the user invokes the session from another region
or network than that usually used.
A number of requirements, motivations and implementation
guidelines for relay service invocation can be found in RFC 3351
[19].
8. Security Considerations
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
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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
[17].
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.
9. 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
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
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Radhika R. Roy
SAIC
3465-B Box Hill Corporate Center Drive
Abingdon, MD 21009
Tel: 443 402 9041
Email: Radhika.R.Roy@saic.com
Henry Sinnreich
pulver.com
115 Broadhollow Rd
Suite 225
Melville, NY 11747
USA
Tel: +1.631.961.8950
Gregg C Vanderheiden
University of Wisconsin-Madison
Trace R & D Center
1550 Engineering Dr (Rm 2107)
Madison, Wi 53706
USA
Phone +1 608 262-6966
FAX +1 608 262-8848
Email: gv@trace.wisc.edu
Arnoud A. T. van Wijk
Viataal
Centre for R & D on sensory and communication disabilities.
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands.
Email: a.vwijk@viataal.nl
10. References
10.1 Normative references
1. S. Bradner, "Intellectual Property Rights in IETF Technology
", BCP 79, RFC 3979, IETF, March 2005.
2. S. Bradner, "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, IETF, 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).
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5. G. Hellstrom, "RTP Payload for Text Conversation", RFC 4103,
IETF, 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,
IETF, June 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, IETF, January 1998.
13. 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)"
14. TIA/EIA/IS-127-2 "Enhanced Variable Rate Codec, Speech Service
Option 3 for Wideband Spread Spectrum Digital Systems. Addendum
2."
15. 3GPP TS26.226 "Cellular Text Telephone Modem Description"
(CTM).
16. H. Sinnreich, S. Lass, and C. Stredicke, "SIP Telephony
Device Requirements and Configuration," IETF, June 2005 - Work in
Progress.
17. Baugher, McGrew, Carrara, Naslund, Norrman, "The Secure Real
Time Transport Protocol (SRTP)", RFC 3711, IETF, March 2004.
18. "IP Multimedia default codecs". 3GPP TS 26.235
19. Charlton, Gasson, Gybels, Spanner, van Wijk, "User
Requirements for the Session Initiation Protocol (SIP) in Support
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of Deaf, Hard of Hearing and Speech-impaired Individuals", RFC
3351, IETF, August 2002.
20. J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with the
Session Description Protocol (SDP)", RFC 3624, IETF, June 2002.
21. ITU-T Recommendation F.700,"Framework Recommendation for
Multimedia Services", November 2000.
22. H. Schulzrinne, S.Casner, R. Frederick, V. Jacobsone, "RTP: A
Transport Protocol for Real-Time Applications", RFC 3550, IETF,
July 2003.
23. ITU-T Recommendation F.703,"Multimedia Conversational
Services", November 2000.
24. J. Rosenberg, H. Schulzrinne, P. Kyzivat, "Indicating User
Agent Capabilities in the Session Initiation Protocol (SIP)", RFC
3840, IETF, August 2004
10.2 Informative references
I. A relay service allows the users to transcode between different
modalities or languages. In the context of this document, relay
services will often refer to text relays that transcode text into
voice and vice-versa. See for example http://www.typetalk.org.
II. 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. 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.)
IV. International Telecommunication Union (ITU), "600/1200-baud
modem standardized for use in the general switched telephone
network. ITU-T Recommendation V.23, November 1988.
Intellectual Property Statement
The IETF takes no position regarding the validity or scope of any
Intellectual Property Rights or other rights that might be claimed
to pertain to the implementation or use of the technology
described in this document or the extent to which any license
under such rights might or might not be available; nor does it
represent that it has made any independent effort to identify any
such rights. Information on the procedures with respect to rights
in RFC documents can be found in BCP 78 and BCP 79.
Copies of IPR disclosures made to the IETF Secretariat and any
assurances of licenses to be made available, or the result of an
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attempt made to obtain a general license or permission for the use
of such proprietary rights by implementers or users of this
specification can be obtained from the IETF on-line IPR repository
at http://www.ietf.org/ipr.
The IETF invites any interested party to bring to its attention
any copyrights, patents or patent applications, or other
proprietary rights that may cover technology that may be required
to implement this standard. Please address the information to the
IETF at ietf-ipr@ietf.org.
Disclaimer of Validity
This document and the information contained herein are provided on
an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE
REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND
THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES,
EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT
THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR
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PARTICULAR PURPOSE.
Copyright Statement
Copyright (C) The Internet Society (2005). This document is
subject to the rights, licenses and restrictions contained in BCP
78, and except as set forth therein, the authors retain all their
rights.
Acknowledgment
Funding for the RFC Editor function is currently provided by the
Internet Society.
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