Text over IP


Text over IP is a means of providing a real-time text service that operates over IP-based networks. It complements Voice over IP and Video over IP.
Real-time text is streaming text that is transmitted as it is produced, allowing text to be used conversationally. Real-time text is defined in ITU-T Multimedia Recommendation F.700 2.1.2.1. Real-time text is designed for conversational use where people interactively converse with each other. To achieve this, particular user requirements have been specified for the delay of each character and the character loss rate.
Real-time Text over IP can be used:
ToIP is designed around the ITU-T T.140 real-time text presentation layer protocol. T.140 allows real-time editing of text e.g. by using 'backspace' and retyping. T.140 is based on the ISO 10646-1 character set that is used by most IP text specifications and uses the UTF-8 format.
Transport of ToIP uses the same Real-time Transport Protocol as VoIP and Video-over-IP. The text is encoded according to IETF RFC 4103 "RTP Payload for Text Conversation".
RFC 4103 supports an optional forward error correction scheme based on redundant transmission. This results in a very low end-to-end packet loss across IP transmission links that have moderately high packet loss. To improve efficiency, text can be buffered for 0.3 – 0.5 seconds before it is sent whilst still meeting the delay requirements.
RTP is usually transported over the User Datagram Protocol. However, because 2.5G mobile networks supported the Transmission Control Protocol but did not consistently support UDP, some implementations of ToIP over mobile networks use TCP internally. 3G mobile networks can support UDP.
The protocol stack for a ToIP medium is:
Very fast typing results in a two kilobit per second traffic load.
Control of ToIP sessions has been defined using the standard Session Initiation Protocol and the Session Description Protocol protocols.
See IETF RFC 5194 "Framework for real-time text over IP using the Session Initiation Protocol " and IETF RFC 4504 "SIP Telephony Device Requirements and Configuration" Section 2.9 for more information.

Deployment

is a concept developed by telecommunication service providers and their suppliers. It aims to create a true multi-service network based in IP technology.
ToIP has been specified for inclusion in the 3GPP IP Multimedia Subsystem . IMS is being used to implement NGNs in many fixed and mobile networks.
Support of ToIP is being considered in multimedia Emergency Public-safety answering point in Europe and USA. The ECRIT IETF working group defines ToIP as one form of access to Emergency Services.
ToIP can provide a 'low impact' solution to meeting national regulatory requirements to provide 'equivalent service' to the telephone service for people who have hearing or speech impairments.
A typical terminal on a fixed line access is a home computer that supports multimedia communications - Voice and Video and real-time Text over IP. See External links for information about ToIP equipment and software.

Use by Deaf and hard-of-hearing people

or TTYs were designed to transport real-time text over the PSTN. TDDs use a range of modem technologies.

Text-over-IP has been designed as a replacement for TDDs when using the IP-based networks but also to be of use to mainstream voice call users. It has less service restrictions compared with TDDs, is designed to be used as a mainstream service and can be used on standard computers or mobile terminals. Proper alerting systems for incoming calls need to be included as well as user interfaces, both hardware and software, that meet the needs of Deaf people, and people with hearing or speech impairments. This can best be achieved with input from end-users in the development stages.
Interworking between TDDs and ToIP has been implemented using gateways by T-Meeting, Omnitor, Trace R& D, RNID, Center, Voiceriver, and AnnieS. RFC 5194 "Framework for real-time text over IP using the Session Initiation Protocol " provides an overview of interworking issues. Work is being proposed in the IETF SIPPING work group on more detailed interworking based on a range of call scenarios.