The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE (VoLTE).
Voice over Internet Protocol (VoIP), also known as IP telephony, refers to a set of technologies used for voice communication sessions over Internet Protocol (IP) networks, such as the Internet. VoIP enables voice calls to be transmitted as data packets, facilitating various methods of voice communication, including traditional applications like Skype, Microsoft Teams, Google Voice, and VoIP phones. Regular telephones can also be used for VoIP by connecting them to the Internet via analog telephone adapters (ATAs), which convert traditional telephone signals into digital data packets that can be transmitted over IP networks.
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.
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 a standardised 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. Various voice over IP technologies are available on smartphones; IMS provides a standard protocol across vendors.
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.
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. This is in contrast to a standard phone which uses the traditional public switched telephone network (PSTN).
Videotelephony is the use of audio and video for simultaneous two-way communication. Today, videotelephony is widespread. There are many terms to refer to videotelephony. Videophones are standalone devices for video calling. In the present day, devices like smartphones and computers are capable of video calling, reducing the demand for separate videophones. Videoconferencing implies group communication. Videoconferencing is used in telepresence, whose goal is to create the illusion that remote participants are in the same room.
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 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 911 refers to an initiative aimed at updating the 911 service infrastructure in the United States and Canada to improve public emergency communications services in a growing wireless mobile society. In addition to calling 911 from a phone, it intends to enable the public to transmit text, images, video and data to the 911 center. The initiative also envisions additional types of emergency communications and data transfer. This NG911 infrastructure is intended to replace the current services over time. The National Emergency Number Association (NENA) first identified the need for NG911 in 2000, and started development actions in 2003, and is nearing full definition and standards for NG911. 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 NG911. On January 24, 2013, the CRTC announced the first step toward a Canadian implementation of NG911 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 NG911, as of October 2013.
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.
MicroSIP is a lightweight, open-source software application that enables users to make Voice over IP (VoIP) calls using the SIP standard. It is designed to be simple, easy to use, and resource-efficient, making it an ideal choice for low-resource devices, such as older computers, and for users who need a minimalistic VoIP client. MicroSIP allows users to initiate audio and video calls, as well as exchange messages, via SIP-based VoIP networks.
The Session Initiation Protocol (SIP) is the signaling protocol selected by the 3rd Generation Partnership Project (3GPP) to create and control multimedia sessions with multiple participants in the IP Multimedia Subsystem (IMS). It is therefore a key element in the IMS framework.
