3pcc

Last updated

In the traditional telephony context, third party call control allows one entity (which we call the controller) to set up and manage a communications relationship or telephone call between two or more other parties. Third party call control (referred to as 3pcc) is often used for operator services (where an operator creates a call that connects two participants together) and conferencing.

Similarly, many SIP services are possible through third party call control. These include the traditional ones on the PSTN, but also new ones such as click-to-dial. Click-to-dial allows a user to click on a web page to speak to a customer service representative. The web server then creates a call between the user and a customer service representative. The call can be between two phones, a phone and an IP host, or two IP hosts.

Third party call control is possible using only the mechanisms specified within RFC   3725. Indeed, many different call flows are possible, each of which will work with SIP compliant user agents. However, there are benefits and drawbacks to each of these flows. The usage of third party call control also becomes more complex when aspects of the call utilize SIP extensions or optional features of SIP. In particular, the usage of RFC   3312 (used for coupling of signaling to resource reservation) with third party call control is non-trivial, and is discussed in Section 9 of RFC 3725. Similarly, the usage of early media (where session data is exchanged before the call is accepted) with third party call control is not trivial; both of them specify the way in which user agents generate and respond to SDP, and it is not clear how to do both at the same time. This is discussed further in Section 8.

Third Party Call Control is perfectly suited for delivery of 9-1-1 calls in a Next Generation 9-1-1 environment. It allows for re-connection of the media stream to an alternate end point if the original destination end point becomes unavailable without dropping the original emergency call (which has a separate media stream to the 3PCC server). Additionally, it minimizes bandwidth requirements between the 3PCC server and the emergency call handling end point because all mixed media uses the same RTP stream. Finally, it does not rely on a destination end point for end to end reliability but rather on a more robust centralized data center with several redundantly deployed servers.

See also

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).

SIMPLE, the Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions, is an instant messaging (IM) and presence protocol suite based on Session Initiation Protocol (SIP) managed by the Internet Engineering Task Force. Contrary to the vast majority of IM and presence protocols used by software deployed today, SIMPLE is an open standard like XMPP.

Telephone number mapping is a system of unifying the international telephone number system of the public switched telephone network with the Internet addressing and identification name spaces. Internationally, telephone numbers are systematically organized by the E.164 standard, while the Internet uses the Domain Name System (DNS) for linking domain names to IP addresses and other resource information. Telephone number mapping systems provide facilities to determine applicable Internet communications servers responsible for servicing a given telephone number using DNS queries.

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.

STUN is a standardized set of methods, including a network protocol, for traversal of network address translator (NAT) gateways in applications of real-time voice, video, messaging, and other interactive communications.

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.

H.248

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.

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. This is in contrast to a standard phone which uses the traditional public switched telephone network (PSTN).

Interactive Connectivity Establishment (ICE) is a technique used in computer networking to find ways for two computers to talk to each other as directly as possible in peer-to-peer networking. This is most commonly used for interactive media such as Voice over Internet Protocol (VoIP), peer-to-peer communications, video, and instant messaging. In such applications, communicating through a central server would be slow and expensive, but direct communication between client applications on the Internet is very tricky due to network address translators (NATs), firewalls, and other network barriers.

A back-to-back user agent (B2BUA) is a logical network element in Session Initiation Protocol (SIP) applications. SIP is a signaling protocol for managing multimedia Voice over Internet Protocol (VoIP) telephone calls. A back-to-back user agent operates between both end points of a communications session and divides the communication channel into two call legs, and mediates all SIP signaling between the endpoints of the session, from establishment to termination. As all control messages for each call flow through the B2BUA, a service provider may implement value-added features available during the call.

The SIP URI scheme is a Uniform Resource Identifier (URI) scheme for the Session Initiation Protocol (SIP) multimedia communications protocol. A SIP address is a URI that addresses a specific telephone extension on a voice over IP system. Such a number could be a private branch exchange or an E.164 telephone number dialled through a specific gateway. The scheme was defined in RFC 3261.

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.

The Media Server Control Markup Language (MSCML) is a protocol used in conjunction with the Session Initiation Protocol (SIP) to enable the delivery of advanced multimedia conferencing services over IP networks. The MSCML specification has been published by the IETF under RFC 4722, now obsoleted by the newer RFC 5022. MSCML was pioneered by the media server company Snowshore, now part of the Dialogic Corporation. MSCML built on ideas from the Netann protocol, and in turn inspired the MSML. An IETF working group called MediaCTRL have now embarked on a standardization of media server scripting languages, drawing on these earlier efforts. Voice scripting protocols like VoiceXML and CCXML are also inspiring sources, and in some cases need to be integrated with what media servers will need to operate in the real world.

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.

Peer-to-peer SIP (P2P-SIP) is an implementation of a distributed voice over Internet Protocol (VoIP) or instant messaging communications application using a peer-to-peer (P2P) architecture in which session control between communication end points is facilitated with the Session Initiation Protocol (SIP).

SIP trunking is a voice over Internet Protocol (VoIP) technology and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and unified communications facilities. Most unified communications applications provide voice, video, and other streaming media applications such as desktop sharing, web conferencing, and shared whiteboard.

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.

This page is based on this Wikipedia article
Text is available under the CC BY-SA 4.0 license; additional terms may apply.
Images, videos and audio are available under their respective licenses.