SIPPING Workgroup
Internet Draft A. van Wijk
Category: Informational AnnieS
Expires: September 5 2006 March 6, 2006
Framework for real-time text over IP using SIP
draft-ietf-sipping-toip-04.txt
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Abstract
This document provides a framework for the implementation of real-
time text conversation over the IP network using the Session
Initiation Protocol and the Real-Time Transport Protocol. It lists
the essential requirements for real-time Text-over-IP (ToIP) and
defines a framework for implementation of all required functions
based on existing protocols and techniques. This includes
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. Requirements...................................................6
5.1 General requirements for ToIP..............................6
5.2 Detailed requirements for ToIP.............................8
5.2.1 Session control and set-up requirements...............8
5.2.2 Transport requirements................................9
5.2.3 Transcoding service requirements.....................10
5.2.4 Presentation and User control requirements...........11
5.2.5 Interworking requirements............................12
5.2.5.1 PSTN Interworking requirements..................12
5.2.5.2 Cellular Interworking requirements..............12
5.2.5.3 Instant Messaging Interworking requirements.....13
6. Implementation Framework......................................13
6.1 Framework of general implementation.......................13
6.2 Framework of detailed implementation......................14
6.2.1 Session control and set-up...........................14
6.2.1.1 Pre-session setup...............................14
6.2.1.2 Basic Point-to-Point Session setup..............15
6.2.1.3 Addressing......................................15
6.2.1.4 Session Negotiations............................15
6.2.1.5 Additional session control......................16
6.2.2 Transport............................................16
6.2.3 Transcoding services.................................17
6.2.4 Presentation and User control functions..............18
6.2.4.1 Progress and status information.................18
6.2.4.2 Alerting........................................18
6.2.4.3 Answering Machine...............................18
6.2.4.4 Text presentation...............................19
6.2.4.5 File storage....................................19
6.2.5 Interworking functions...............................19
6.2.5.1 PSTN Interworking...............................20
6.2.5.2 Mobile Interworking.............................21
6.2.5.2.1 Cellular "No-gain".........................21
6.2.5.2.2 Cellular Text Telephone Modem (CTM)........21
6.2.5.2.3 Cellular "Baudot mode".....................22
6.2.5.2.4 Mobile data channel mode...................22
6.2.5.2.5 Mobile ToIP................................22
6.2.5.3 Instant Messaging Interworking..................22
6.2.5.4 Interworking through gateways...................23
6.2.5.5 Multi-functional Combination gateways...........24
6.2.5.6 Character set transcoding.......................25
7. Further recommendations for implementers and service providers25
7.1 Access to Emergency services..............................25
7.2 Home Gateways or Analog Terminal Adapters.................26
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7.3 User Mobility.............................................26
7.4 Firewalls and NATs........................................26
8. IANA Considerations...........................................26
9. Security Considerations.......................................26
10. Authors Addresses...........................................27
11. References...................................................28
11.1 Normative references.....................................28
11.2 Informative references...................................30
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) and
Multimedia-over-IP (MoIP) environments, as well as meeting the users
requirements, including those of deaf, hard of hearing and speech-
impaired users as described in RFC3351 [2] and mainstream users.
The Session Initiation Protocol (SIP) [3] 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) [4] is the protocol of choice
for real-time data transmission, and its use for real-time 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 [6].
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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 [6]. It defines the:
a. Requirements of Real-time text;
b. Requirements for ToIP interworking;
c. Description of ToIP implementation 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 [7] and indicate requirement levels for
compliant implementations.
4.
Definitions
Audio bridging: a function of an audio media bridge server, gateway
or relay service that bridges audio into a single source through
combining audio from multiple users excluding each destination
sources audio and sends to each respective destination enabling an
audio path through the service between the users involved in the
call.
Cellular: a telecommunication network that has wireless access and
can support voice and data services over very large geographical
areas. Also called Mobile.
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.
(Equivalent to real-time text.)
Real-time 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.
Conversational text is defined in ITU-T F.700 Framework for
multimedia services [25].
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Text gateway: a function that transcodes between different forms of
real-time text transport methods, e.g., between ToIP in IP networks
and Baudot or ITU-T V.21 text telephony in the PSTN.
Textphone: 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 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
real-time text in a call.
Text Bridging: a function of the text media bridge server, gateway or
relay service that bridges real-time text into a single source
through combining real-time text from multiple users excluding each
destination sources real-time text and sends to each respective
destination enabling a real-time text path through the service
between the users involved in the call.
Text telephony: analog textphone service.
Total Conversation: a multimedia service offering real time
conversation in video, real-time text and voice according to
interoperable standards. All media flow in real time. (See ITU-T
F.703 "Multimedia conversational services" [6].)
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
real-time text and audio.
TTY: 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.
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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 Real-time Text over Internet Protocol
UTF-8 Universal Transfer Format-8
VCO/HCO Voice Carry Over/Hearing Carry Over
VoIP Voice over Internet Protocol
5.
Requirements
This framework defines a text-based conversational service that is
the text equivalent of voice based telephony. This section describes
the requirements that the framework is designed to meet and the
functionality it should offer.
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, speech-impaired and mainstream
users.
This section (Requirements) informs implementers about WHICH
requirements the systems and services shall meet. The next section
(Section 6 Framework Implementation) describes HOW to do it.
5.1
General requirements for ToIP
Any framework for ToIP must be designed to meet the requirements of
RFC3351 [2]. A basic requirement is that it must provide a
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standardized way for offering text-based, conversational services
that can be used as an equivalent to voice telephony by deaf, hard of
hearing speech-impaired and mainstream users.
It is important to understand that real-time text conversations are
significantly different from other text-based communications like
email or Instant Messaging. Real-time text conversations deliver an
equivalent mode to voice conversations by providing transmission of
text character by character as it is entered, so that the
conversation can be followed closely and immediate interaction take
place.
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.
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 transport and presentation of the conversation;
b. Provide simultaneous transmission in both directions;
c. Support both point-to-point and multipoint communication;
d. Allow other media, like audio and video, to be used in
conjunction with ToIP;
e. Ensure that the real-time text service is always available.
Real-time text is a useful subset of Total Conversation defined in
ITU-T F.703 [6]. Users could use multiple modes of communication
during the conversation, either at the same time or by switching
between modes, e.g., between real-time text and audio.
Deaf, hard-of-hearing and mainstream 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 real-time text in conversational
sessions.
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In many of the above examples, text may accompany speech. The text
could be displayed side by side, or in a manner similar to subtitling
in broadcasting environments, or in any other suitable manner. This
could occur with users who are hard of hearing and also for mixed
media calls with both hearing and deaf people participating in the
call.
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.
5.2
Detailed requirements for ToIP
The following sections lists individual requirements for ToIP. Each
requirement has been given a uniquely identifier (R1, R2, etc).
Section 6 (Implementation Framework) describes how to implement ToIP
based on these requirements and using existing protocols and
techniques.
5.2.1
Session control and set-up requirements
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 the real-time text.
Simultaneous or alternating use of voice and real-time 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).
R1: It SHOULD be possible to start conversations in any mode (real-
time text, voice, video) or combination of modes.
R2: It MUST be possible for the users to switch to real-time text, or
add real-time text as an additional modality, during the
conversation.
R3: Systems supporting ToIP MUST allow users to select any of the
supported conversation modes at any time, including mid-conversation.
R4: 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.
R5: If the user requests simultaneous use of real-time text and
audio, and this is not possible either because the system only
supports alternate modalities or because of constraints in the
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network, the system MUST try to establish communication with best
effort.
R6: If the user has expressed a preference for real-time text,
establishment of a connection including real-time text MUST have
priority over other outcomes of the session setup.
R7: It SHOULD be possible to use the real-time text medium in
conference sessions in a similar way to how audio is handled and
video is displayed.
Real-time 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.
R8: During session set up, it SHOULD be possible for the users to
indicate if the caller wants to use voice and real-time text
simutaneously as part of the conversation.
R9: Session set up and negotiation of modalities must allow users to
specify the language of the real-time text to be used. (It is
recommended that similar functionality is provided for the video part
of the conversation, i.e. to specify the sign language being used).
5.2.2
Transport requirements
ToIP will often be used to access a relay service [I], allowing real-
time 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.
R10: Characters must be transmitted soon after entry of each
character so that the maximum delay requirement can be met. A delay
time of one second is regarded good, while a delay of two seconds is
possible to use.
R11: It must be possible to transmit characters at a rate sufficient
to support fast human typing as well as speech to text methods of
generating conversation text. A rate of 20 characters per second is
regarded sufficient.
R12: a ToIP service must be able to deal with international character
sets.
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R13: Where it is possible, loss of real-time text during transport
should be detected and the user should be informed.
R14: Transport of real-time text should be as robust as possible, so
as to minimize loss of characters.
R15: Where possible, it must be possible to send and receive real-
time text simultaneously.
5.2.3
Transcoding service 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.
Some users may indicate their preferred modality to be audio while
others may indicate real-time text. In this case, transcoding
services might be needed for text-to-speech (TTS) and speech-to-text
(STT). Other examples of possible scenarios for including a relay
service in the conversation are: text bridging after conversion from
speech, audio bridging after conversion from real-time text, etc.
A number of requirements, motivations and implementation guidelines
for relay service invocation can be found in RFC 3351 [2].
R16: It MUST be possible for users to invoke a transcoding service
where such service is available.
R17: It MUST be possible for users to indicate their preferred
modality.
R18: The requirements for transcoding services need to be negotiated
in real-time to set up the session.
R19: Adding or removing a relay service MUST be possible without
disrupting the current session.
R20: When setting up a session, it MUST be possible for a user to
determine the type of relay service requested (e.g., speech to text
or text to speech). The specification of a type of relay MUST include
a language specifier.
R21: 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.
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5.2.4
Presentation and User control requirements
R22: User Agents for ToIP services must have alerting methods (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.
R23: Where real-time text is used in conjunction with other media,
exposure of user control functions through the User Interface needs
to be done in an equivalent manner for all supported media.
In other words, where certain call control functions are available
for the audio media part of a session, these functions MUST also be
supported for the real-time text media part of the same session. For
example, call transfer must act on all media in the session.
R24: 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.
R25: When a session invitation involving ToIP originates from a PSTN
text telephone (e.g. transcoded via a text gateway), this SHOULD be
indicated to the user. The ToIP client MAY adjust the presentation of
the real-time text to the user as a consequence.
R26: An indication should be given to the user when real-time text is
available during the call, even if it is not invoked at call setup
(e.g. when only voice and/or video is used initially).
R27: The user MUST be informed of any change in modalities.
R28: Users must be presented with appropriate session progress
information at all times.
R29: Answering machine functions SHOULD be provided by the User
Agent.
R30: When the answering machine function is enabled on the User
Agent, alerting of the user SHOULD still be possible and users SHOULD
be able to take over control from the answering machine function at
any time.
R31: Users SHOULD be able to save the text portion of a conversation.
R32: The presentation of the conversation should be done in such a
way that users can easily identify which party generated any given
portion of text.
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5.2.5
Interworking requirements
There is a range of existing real-time text services. There is also a
range of network technologies that could support real-time text
services.
Real-time/Interactive texting facilities exist already in various
forms and on various networks. On the PSTN, it is commonly referred
to as text telephony.
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.
R33: ToIP SHOULD provide interoperability with text conversation
features in other networks, for instance the PSTN.
R34: 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
interworking network.
R35: Address information, both called and calling, SHOULD be
transferred, and possibly converted, when interworking between
different networks.
R36: When interworking with other networks and services, the ToIP
service SHOULD provide buffering mechanisms to deal with delays in
call setup, transmission speeds and/or to interwork with half duplex
services.
5.2.5.1
PSTN Interworking requirements
Analog text telephony is being used in many countries, mainly by
deaf, hard of hearing and speech-impaired individuals.
R37: ToIP services MUST provide interworking with PSTN legacy text
telephony devices.
R38: When interworking with PSTN legacy text telephony services,
alternating text and voice function MAY be supported. (Called "voice
carry over (VCO) and hearing carry over (HCO)").
5.2.5.2
Cellular Interworking requirements
As mobile communications have been adopted widely, various solutions
for real-time texting while on the move have been developed. ToIP
services should provide interworking with such services as well.
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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 [8] 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.
R39: a ToIP service SHOULD provide interworking with mobile text
conversation services.
5.2.5.3
Instant Messaging Interworking requirements
Many people use Instant Messaging to communicate via the Internet
using text. Instant Messaging usually transfers blocks of text rather
than streaming as is used by ToIP. Usually a specific action is
required by the user to activate transmission, such as pressing the
ENTER key or a send button. 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 [2]. 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.
R39: a ToIP service MAY provide interworking with Instant Messaging
services.
6.
Implementation Framework
This section describes an implementation framework for ToIP that
meets the requirements and offers the functionality as set out in
section 5. The framework presented here uses existing standards that
are already commonly used for voice based conversational services on
IP networks.
6.1
Framework of general implementation
ToIP uses the Session Initiation Protocol (SIP) [3] to set up,
control and tear down the connections between users whilst the media
is transported using the Real-Time Transport Protocol (RTP) [4] as
described in RFC4103 [5].
SIP [3] allows participants to negotiate all media including real-
time text conversation [5]. This is a highly desirable function for
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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. Even for mainstream users, media negotiations like real-
time text are also very useful in many circumstances as described
earlier.
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.
Real-time text conversation based on the presentation protocol T.140
[9], 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 basic real-time editing of the text.
6.2
Framework of detailed implementation
6.2.1
Session control and set-up
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) 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.
6.2.1.1
Pre-session setup
The requirements of the user to be reached at a consistent address
and to store preferences for evaluation at session setup are met by
pre-session setup actions. That includes 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.
The need to use real-time 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 real-time text by registering that preference with a
SIP server that is part of the ToIP service.
b. Server support of User Preferences: SIP servers that support ToIP
services MUST have the capability to act on calling user preferences
for real-time text in order to accept or reject the session.The
actions taken can be based on the called users preferences defined
as part of the pre-session setup registration. For example, if the
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user is called by another party, and it is determined that a
transcoding server is needed, the session should be re-directed or
otherwise handled accordingly.
6.2.1.2
Basic Point-to-Point Session setup
A point-to-point session takes place between two parties. For ToIP,
one or both of the communicating parties will indicate real-time text
as a possible or preferred medium for conversation using SIP in the
session setup.
The following features MAY be implemented to facilitate the session
establishment using ToIP:
a. Caller Preferences: SIP headers (e.g., Contact)[11] can be used to
show that ToIP is the medium of choice for communications.
b. Called Party Preferences [12]: The called party being passive can
formulate a clear rule indicating how a session should be handled
either using real-time 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: It is RECOMMENDED that
SIP servers also handle the incoming sessions in accordance with
preferences expressed for real-time text. The SIP Server can also
enforce ToIP policy rules for communications (e.g. use of the
transcoding server for ToIP).
6.2.1.3
Addressing
The SIP [3] addressing schemes MUST be used for all entities in a
ToIP session. For example, SIP URLs or Tel URLs are used for
caller, called party, user devices, and servers (e.g., SIP server,
Transcoding server).
6.2.1.4
Session Negotiations
The Session Description Protocol (SDP) used in SIP [3] provides the
capabilities to indicate real-time text as a medium in the session
setup. RFC 4103 [5] uses the RTP payload types "text/red" and
"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, SIPs offer/answer model [13] can also be used in
conjunction with other capabilities including the use of a
transcoding server for enhanced session negotiations [14,15,16].
Systems SHOULD provide a best-effort approach to answering
invitations for session set-up and users SHOULD be informed when the
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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.
6.2.1.5
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.
6.2.2
Transport
A ToIP service MUST always support at least one real-time text media
type.
ToIP services MUST support the Real-Time Transport Protocol (RTP) [4]
according to the specification of RFC4103 [4] for the transport of
text between participants.
RFC4103 describes the transmission of T.140 [9] real-time text on IP
networks.
In order to enable the use of international character sets, the
transmission format for text conversation SHALL be UTF-8 [17], in
accordance with ITU-T T.140.
If real-time text is detected to be missing after transmission, there
SHOULD be a "text loss" indication in the real-time text as specified
in T.140 Addendum 1 [9].
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 real-time text transmission. A
redundancy level using 2 generations gives very reliable results and
is therefore strongly RECOMMENDED.
Real-time text capability MUST be announced in SDP by a declaration
similar to this example:
m=text 11000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
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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.
Real-time 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.
The text transmission SHALL allow a rate of at least 30 characters
per second.
6.2.3
Transcoding services
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.
A specific type of transcoding service in a ToIP environment is a
relay service. 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 real-
time text and real-time 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.
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 [14] 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.
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6.2.4
Presentation and User control functions
6.2.4.1
Progress and status 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 real-time text, voice and video channels
- Session progress
- Incoming real-time text
- Any loss in incoming real-time text
- Typed and transmitted real-time text.
6.2.4.2
Alerting
For users who cannot use the audible alerter for incoming sessions,
it is RECOMMENDED to include a tactile as well as a visual indicator.
Among the alerting options are alerting by the User Agents User
Interface and specific alerting user agents registered to the same
registrar as the main user agent.
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.
6.2.4.3
Answering Machine
Systems for ToIP MAY support an answering machine function,
equivalent to answering machines on telephony networks. If an
answering machine function is supported, it MUST support at least 160
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characters for the greeting message. It MUST support incoming real-
time 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.
6.2.4.4
Text presentation
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. This could be done using color, positioning of the
text (i.e. incoming real-time text and outgoing real-time text in
different display areas), by in-band identifiers of the parties or by
a combination of any of these techniques.
ToIP SHOULD handle characters such as new line, erasure and alerting
during a session as specified in ITU-T T.140 [9].
6.2.4.5
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 XHTML format [18] including timestamps, party names (or
addresses) and the text conversation.
6.2.5
Interworking functions
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.
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 at the "protocol" level (where
an indication must be sent back to the Gateway). However, the special
care needs to be taken to provide the best possible real-time
performance.
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6.2.5.1
PSTN 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
[19], 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 [II] 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 [III]
might be the first choice.
In particular transmission of interactive text on PSTN networks takes
place using a variety of codings and modulations, including ITU-T
V.21 [III], Baudot [II], 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 [19] standard was developed to
address some of these issues.
ITU-T V.18 [19] 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.
Text gateways MUST use the ITU-T V.18 [19] standard at the PSTN side.
A text gateway MUST act as a SIP User Agent on the IP side and
support RFC4103 text transport.
PSTN-ToIP gateways MUST allow alternating use of real-time 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.
While ToIP allows receiving and sending real-time text simultaneously
and is displayed on a split screen, many analog text telephones
require users to take turns typing.
This is because many text telephones operate strictly half duplex.
Only one can transmit text at a time. The users apply strict turn-
taking rules.
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There are several text telephones which communicate in full duplex,
but merge transmitted text and received text in the same line in the
same display window. And also here do the users apply strict turn
taking rules.
Native V.18 text telephones support full duplex and separate display
from reception and transmission so that the full duplex capability
can be used fully. Such devices could use the ToIP split screen as
well, but almost all text telephones use a restricted character set
and many use low text transmission speeds (4 to 7 charcters per
second).
That is why it is important for the ToIP user to know that he or she
is connected with an analog text telephone. The "txp" media content
attribute [10]SHOULD be used to indicate that the call originates
from a PSTN text telephone (e.g. via an ATA or a text gateway).
6.2.5.2
Mobile Interworking
Mobile wireless (or Cellular) 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.
ToIP may be supported over the cellular wireless packet switched
service. It interfaces to the Internet.
The following sections describe how mobile text telephony is
supported.
6.2.5.2.1
Cellular "No-gain"
The "No-gain" text telephone transporting technology uses specially
modified EFR [20] and EVR [21] 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.
6.2.5.2.2
Cellular Text Telephone Modem (CTM)
CTM [8] 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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6.2.5.2.3
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. Thus it is a common name
for the "No-Gain" and the CTM solutions when applied to the Baudot
type textphones.
6.2.5.2.4
Mobile data channel mode
Many mobile terminals allow the use of the circuit switched data
channel to transfer data in real-time. Data rates of 9600 bit/s are
usually supported on the 2G mobile network. Gateways provide
interoperability with PSTN textphones.
6.2.5.2.5
Mobile ToIP
ToIP could be supported over mobile wireless packet switched services
that interface to the Internet. For 3GPP 3G services, ToIP support is
described in 3G TS 26.235 [22].
6.2.5.3
Instant Messaging Interworking
Text gateways MAY be used 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 like ToIP, gateways MUST transcode between the two
formats. Text gateways for interworking between Instant Messaging and
ToIP MUST apply a procedure for bridging the different conversational
formats of real-time text versus text messaging. The following advice
may improve user experience for both parties in a call through a
messaging gateway.
a. Concatenate individual characters originating at the ToIP side
into blocks of text.
b. 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.
c. 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.
d. 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.
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e. Text Gateways must be capable to maintain the real-time
performance for ToIP while providing the interworking services.
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 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 can be the
same for Instant Messaging (SIMPLE) and ToIP. Thus, the user may
simply use different applications for ToIP and text messaging in the
same terminal.
Devices that implement Instant Messaging SHOULD implement ToIP as
described in this document so that a more complete text communication
service can be provided.
6.2.5.4
Interworking through gateways
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.
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.
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.
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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 3000 characters).
ToIP interworking requires a method to invoke a text gateway. As
described previously, these text gateways MUST act as User Agents at
the IP side. The capabilities of the 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 real-time text, so the 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 real-time text subscriptions, linked to the phone number
or terminal identifier/ IP address.
c. Real-time text capability indicators.
d. Real-time 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 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.
6.2.5.5
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 ToIP MUST support the following interworking functions:
- PSTN text telephony to ToIP.
- PSTN text telephony to Instant Messaging.
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- Instant Messaging to ToIP.
6.2.5.6
Character set 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, where this character
exists. For legacy systems where the character does not exist, the
. (full stop) character SHOULD be used instead.
7.
Further recommendations for implementers and service providers
7.1
Access to 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 real-time text medium MUST be
equivalent to the emergency service provided to users utilising
speech or other media.
A text gateway MUST be able to route real-time 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 real-time
text calls to emergency services MAY require the use of a transcoding
service.
A text gateway with cellular wireless packet switched services MUST
be able to route real-time text calls to emergency service providers
when any of the recognized emergency numbers that support real-time
text communication for the country is called.
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7.2
Home Gateways or Analog Terminal Adapters
Analog terminal adapters (ATA) using SIP based IP communication and
RJ-11 connectors for connecting traditional PSTN devices SHOULD
enable connection of legacy PSTN text telephones [23].
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 [4], in a similar
way as it provides interoperability for voice sessions.
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.
7.3
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.
7.4
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.
8.
IANA Considerations
There are no IANA considerations for this specification.
9.
Security Considerations
User confidentiality and privacy need to be met as described in SIP
[3]. For example, nothing should reveal the fact that the ToIP user
might be a person with a hearing or speech impairment. ToIP is after
all a mainstream communication medium for all users. It is up to the
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ToIP user to make his or her hearing or speech impairment 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 [24].
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.
10.
Authors Addresses
The following people provided substantial technical and writing
contributions to this document, listed alphabetically:
Willem Dijkstra
TNO Informatie- en Communicatietechnologie
Eemsgolaan 3
9727 DW Groningen
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: btdingle@gmail.com
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 7608 0511
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
Foundation for an Information and Communication Network for the Deaf
and Hard of Hearing
"AnnieS"
www.annies.nl
Email: arnoud@annies.nl
11.
References
11.1
Normative references
1. S. Bradner, "Intellectual Property Rights in IETF Technology",
BCP 79, RFC 3979, IETF, March 2005.
2. 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.
3. J. Rosenberg, H. Schulzrinne, G. Camarillo, A. R. Johnston, J.
Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: Session
Initiation Protocol", RFC 3621, IETF, June 2002.
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4. H. Schulzrinne, S.Casner, R. Frederick, V. Jacobsone, "RTP: A
Transport Protocol for Real-Time Applications", RFC 3550, IETF,
July 2003.
5. G. Hellstrom, P. Jones, "RTP Payload for Text Conversation", RFC
4103, IETF, June 2005.
6. ITU-T Recommendation F.703,"Multimedia Conversational Services",
November 2000.
7. S. Bradner, "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, IETF, March 1997
8. 3GPP TS 26.226 "Cellular Text Telephone Modem Description"
(CTM).
9. ITU-T Recommendation T.140, "Protocol for Multimedia Application
Text Conversation" (February 1998) and Addendum 1 (February
2000).
10. J. Hautakorpi, G. Camarillo, "The SDP (Session Description
Protocol) Content Attribute", IETF, February 2006 - Work in
Progress.
11. J. Rosenberg, H. Schulzrinne, P. Kyzivat, "Indicating User Agent
Capabilities in the Session Initiation Protocol (SIP)", RFC
3840, IETF, August 2004
12. J. Rosenberg, H. Schulzrinne, P. Kyzivat, "Caller Preferences
for the Session Initiation Protocol (SIP)", RFC 3841, IETF,
August 2004
13. J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with the
Session Description Protocol (SDP)", RFC 3624, IETF, June 2002.
14. G. Camarillo, "Framework for Transcoding with the Session
Initiation Protocol" IETF Nov 2005 - Work in progress.
15. 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.
16. G. Camarillo, "The SIP Conference Bridge Transcoding Model,"
IETF, Jan 2006 - Work in Progress.
17. Yergeau, F., "UTF-8, a transformation format of ISO 10646", RFC
3629, IETF,November 2003.
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18. "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.
19. ITU-T Recommendation V.18,"Operational and Interworking
Requirements for DCEs operating in Text Telephone Mode,"
November 2000.
20. 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)"
21. TIA/EIA/IS-127-2 "Enhanced Variable Rate Codec, Speech Service
Option 3 for Wideband Spread Spectrum Digital Systems. Addendum
2."
22. "IP Multimedia default codecs". 3GPP TS 26.235
23. H. Sinnreich, S. Lass, and C. Stredicke, "SIP Telephony Device
Requirements and Configuration," IETF, October 2005 - Work in
Progress.
24. Baugher, McGrew, Carrara, Naslund, Norrman, "The Secure Real
Time Transport Protocol (SRTP)", RFC 3711, IETF, March 2004.
25. ITU-T Recommendation F.700,"Framework Recommendation for
Multimedia Services", November 2000.
11.2
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. 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.)
III. 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.
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.
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