The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.
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 called IP telephony, is a method and group of technologies for voice calls, the delivery of voice communication sessions over Internet Protocol (IP) networks, such as the Internet.
A media gateway is a translation device or service that converts media streams between disparate telecommunications technologies such as POTS, SS7, Next Generation Networks or private branch exchange (PBX) systems. Media gateways enable multimedia communications across packet networks using transport protocols such as Asynchronous Transfer Mode (ATM) and Internet Protocol (IP).
Inter-Asterisk eXchange (IAX) is a communications protocol native to the Asterisk private branch exchange (PBX) software, and is supported by a few other softswitches, PBX systems, and softphones. It is used for transporting VoIP telephony sessions between servers and to terminal devices.
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. 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 that provides such standardization.
The Gateway Control Protocol is an implementation of the media gateway control protocol architecture for providing telecommunication services across a converged internetwork consisting of the traditional public switched telephone network (PSTN) and modern packet networks, such as the Internet. H.248 is the designation of the recommendations developed by the ITU Telecommunication Standardization Sector (ITU-T) and Megaco is a contraction of media gateway control protocol used by the earliest specifications by the Internet Engineering Task Force (IETF). The standard published in March 2013 by ITU-T is entitled H.248.1: Gateway control protocol: Version 3.
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.
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).
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.
Image Share is a service for sharing images between users during a mobile phone call. It has been specified for use in a 3GPP-compliant cellular network by the GSM Association in the PRD IR.79 Image Share Interoperability Specification.
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.
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.
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 with each other, or choose to communicate with any combination of those three modes and to do so in real-time.
Multipath TCP (MPTCP) is an ongoing effort of the Internet Engineering Task Force's (IETF) Multipath TCP working group, that aims at allowing a Transmission Control Protocol (TCP) connection to use multiple paths to maximize throughput and increase redundancy.
IMS is a set of specifications to offer multimedia services through IP protocol. This makes it possible to incorporate all kinds of services, such as voice, multimedia and data, on an accessible platform through any Internet connection.
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.
ViLTE, an acronym for "Video over LTE", is a conversational video service based on the IP Multimedia Subsystem (IMS) core network like VoLTE. It has specific profiles for the control and VoLTE of the video service and uses LTE as the radio access medium. The service as a whole is governed by the GSM Association in PRD IR.94.
