Total conversation

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Total conversation is an ITU standard of simultaneous video, voice and text service in telecommunications. Total conversation allows people in two or more locations to: (a) see each other, (b) hear each other, and (c) conduct a text interaction (real-time text) with each other, or choose to communicate with any combination of those three modes and to do so in real-time. [1]

Contents

The standard was defined in ITU-T recommendation F.703 [2] of 2000 as "an audiovisual conversation service providing bidirectional symmetric real-time transfer of motion video, text and voice between users in two or more locations".

Application

Universal design

Total conversation is a straightforward application of universal design principles applied to telecommunication field. This standard brings communication services to the broadest range of population including deaf people, hard or hearing people but also people who have speech impairment, people with cognitive impairment, elderly people, children, and also ... regular users. It is an enhancement of call services that provides a better experience to all. Total conversation uses video, text and audio. These media can be combined or not depending on the type of profile used : many user profiles have to be taken into account. Such methodology is named Universal Design.

Implementation of the standard

Available implementations

A number of European companies including 4CTel, Aupix, IVèS, nWise, Omnitor, and Orange Labs produced implementations. These are mostly used to provide user-to-user calls and relay services for deaf and hard of hearing people.

REACH112

Total conversation was piloted in a European project called REACH112, [3] named after the 1-1-2 European emergency number, whose purpose was to carry out test deployment of total conversation in a live environment with several thousands users and for three usages:

Specifications

Total conversation specifications describe:

Total conversation defines three basic services

Example of technical protocols used by total conversation are:

(* Use of Common Alerting Protocol may be also considered to share information with Emergency Services.)

Over instant messaging

AOL Instant Messenger supports the simultaneous use of audio, video, and real-time text via the "Real-Time IM" [8] feature. This would be considered compliant with Total Conversation.

For other chat networks such as Google Talk, the XMPP protocol technically allow for total conversation by combining multiple XMPP Extensions, such as XEP-0266 audio, [9] XEP-0299 video, [10] and XEP-0301 real-time text. [11] As of December 2011, Google Talk only supports audio/video, but not real-time text yet.

Skype has been supporting simultaneous audio, video, and real-time text.

IP multimedia subsystem

Total conversation basically provides a multi-media call services with specific media enabled. The currently most common implementation environment is native SIP as described above.

Total conversation can also be implemented in other call control environments. One such environment is IP Multimedia Subsystem (IMS) in its IMS multimedia telephony service.

Firstly, total conversation can be defined as a pragmatic selection of existing standards related to Session Initiation Protocol with the selection of audio, video and text codecs. It is a private initiative of services provider and technology vendors seeking interoperability in the broadtest sense.

Secondly, this can apply to existing and deployed SIP platform rather than future or next generation networks.

Thirdly, IMS is much more ambitious and define a whole architecture, including all internal interfaces and billing and physical infrastructure. These standards are more suitable for large telecommunication operators and their equipment vendors. This complexity has consequences:

Related Research Articles

The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications. SIP is used for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, in instant messaging over Internet Protocol (IP) networks as well as mobile phone calling over LTE (VoLTE).

Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services over the Internet, rather than via the public switched telephone network (PSTN), also known as plain old telephone service (POTS).

Telecommunications device for the deaf

A telecommunications device for the deaf (TDD) is a teleprinter, an electronic device for text communication over a telephone line, that is designed for use by persons with hearing or speech difficulties. Other names for the device include teletypewriter (TTY), textphone, and minicom.

Telecommunications relay service

A telecommunications relay service, also known as TRS, relay service, or IP-relay, or Web-based relay service, is an operator service that allows people who are deaf, hard of hearing, deafblind, or have a speech disorder to place calls to standard telephone users via a keyboard or assistive device. Originally, relay services were designed to be connected through a TDD, teletypewriter (TTY) or other assistive telephone device. Services gradually have expanded to include almost any real-time text capable technology such as a personal computer, laptop, mobile phone, PDA, and many other devices. The first TTY was invented by deaf scientist Robert Weitbrecht in 1964. The first relay service was established in 1974 by Converse Communications of Connecticut.

The IP Multimedia Subsystem or IP Multimedia Core Network Subsystem (IMS) is an architectural framework for delivering IP multimedia services. Historically, mobile phones have provided voice call services over a circuit-switched-style network, rather than strictly over an IP packet-switched network. Alternative methods of delivering voice (VoIP) or other multimedia services have become available on smartphones, but they have not become standardized across the industry. IMS is an architectural framework to provide such standardization.

A session border controller (SBC) is a network element deployed to protect SIP based voice over Internet Protocol (VoIP) networks.

PacketCable network is a technology specification defined by the industry consortium CableLabs for using Internet Protocol (IP) networks to deliver multimedia services, such as IP telephony, conferencing, and interactive gaming on a cable television infrastructure.

The next-generation network (NGN) is a body of key architectural changes in telecommunication core and access networks. The general idea behind the NGN is that one network transports all information and services by encapsulating these into IP packets, similar to those used on the Internet. NGNs are commonly built around the Internet Protocol, and therefore the term all IP is also sometimes used to describe the transformation of formerly telephone-centric networks toward NGN.

VoIP phone

A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet, instead of the traditional public switched telephone network (PSTN).

Videotelephony Reception and transmission of audio-video signals by users at different locations

Videotelephony comprises the technologies for the reception and transmission of audio-video signals by users in different locations, for communication between people in real time as well as showing on your account that you're offline even though you could still chat and voice call . A videophone is a telephone with a video display, capable of simultaneous video and audio for communication between people in real time. Videoconferencing implies the use of this technology for a group or organizational meeting rather than for individuals, in a videoconference. Telepresence may refer either to a high-quality videotelephony system or to meetup technology, which goes beyond video into robotics. Videoconferencing has also been called "visual collaboration" and is a type of groupware.

Video relay service Video telecommunication service

A video relay service (VRS), also sometimes known as a video interpreting service (VIS), is a video telecommunication service that allows deaf, hard-of-hearing, and speech-impaired (D-HOH-SI) individuals to communicate over video telephones and similar technologies with hearing people in real-time, via a sign language interpreter.

Text over IP is a means of providing a real-time text (RTT) service that operates over IP-based networks. It complements Voice over IP (VoIP) and Video over IP.

H.323 is a Recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences.

The Universal Mobile Telecommunications System (UMTS) is one of the new ‘third generation’ 3G mobile cellular communication systems. UMTS builds on the success of the ‘second generation’ GSM system. One of the factors in the success of GSM has been its security features. New services introduced in UMTS require new security features to protect them. In addition, certain real and perceived shortcomings of GSM security need to be addressed in UMTS.

Video Share

Video Share is an IP Multimedia System (IMS) enabled service for mobile networks that allows users engaged in a circuit switch voice call to add a unidirectional video streaming session over the packet network during the voice call. Any of the parties on the voice call can initiate a video streaming session. There can be multiple video streaming sessions during a voice call, and each of these streaming sessions can be initiated by any of the parties on the voice call. The video source can either be the camera on the phone or a pre-recorded video clip.

The 3GPP/NGN IP Multimedia Subsystem (IMS) multimedia telephony service (MMTel) is a global standard based on the IMS, offering converged, fixed and mobile real-time multimedia communication using the media capabilities such as voice, real-time video, text, file transfer and sharing of pictures, audio and video clips. With MMTel, users have the capability to add and drop media during a session. You can start with chat, add voice, add another caller, add video, share media and transfer files, and drop any of these without losing or having to end the session. MMTel is one of the registered ICSI feature tags.

Next Generation 9-1-1 refers to an initiative aimed at updating the 9-1-1 service infrastructure in the United States and Canada to improve public emergency communications services in a growingly wireless mobile society. In addition to calling 9-1-1 from a phone, it intends to enable the public to transmit text, images, video and data to the 9-1-1 center. The initiative also envisions additional types of emergency communications and data transfer. This NG9-1-1 infrastructure is intended to replace the current services over time. The National Emergency Number Association (NENA) first identified the need for NG9-1-1 in 2000, and started development actions in 2003, and is nearing full definition and standards for NG9-1-1. Since 2006, the US Department of Transportation (DOT) in the United States and the Canadian Radio-television and Telecommunications Commission (CRTC) in Canada have been leading their respective initiatives, which include research and development projects aimed at advancing NG9-1-1. On January 24, 2013, the CRTC announced the first step toward a Canadian implementation of NG9-1-1 and, in March 2016, began a consultation with the public to discuss what services should be offered, who will play a role in offering these services and how these services should be paid for. Several US states have implemented versions of NG9-1-1, as of October 2013.

Real-time text Text transmitted instantly as it is typed or created

Real-time text (RTT) is text transmitted instantly as it is typed or created. Recipients can immediately read the message while it is being written, without waiting.

The Session Initiation Protocol (SIP) is the signaling protocol selected by the 3rd Generation Partnership Project (3GPP) to create and control multimedia sessions with two or more participants in the IP Multimedia Subsystem (IMS), and therefore is a key element in the IMS framework.

Assistive Technology for the Deaf and Hard of Hearing is special technology made to assist them including Hearing aids, Video relay services, tactile devices, alerting devices and technology for supporting communication.

References

  1. Total Conversation explained at realtimetext.org
  2. F.703 : Multimedia conversational services
  3. Reach112 project website
  4. Next Generation 112 Archived 2011-09-29 at the Wayback Machine
  5. T.140 : T.140
  6. RFC4103 RTP Payload for T.140
  7. RTT in SIP: Text over IP (ToIP)
  8. AOL Instant Messenger "Real-Time IM" Archived 2012-03-12 at the Wayback Machine
  9. XEP-0266 Codecs for Jingle Audio
  10. XEP-0299 Codecs for Jingle Video
  11. XEP-0301 In-Band Real-Time Text
  12. IMS Reality check
  13. MMTel does anyone other than Ericsson care?
  14. RIch Communication Suite