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Back-to-back User Agent
A back-to-back user agent (B2BUA) is a logical network element in Session Initiation Protocol (SIP) applications., ''SIP: Session Initiation Protocol'', IETF, The Internet Society (2002) 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. In the originating call leg, the B2BUA acts as a user agent server (UAS) and processes the request as a user agent client (UAC) to the destination end, handling the signaling between end points back-to-back. A B2BUA maintains complete state for the calls it handles. Each side of a B2BUA operates as a standard SI ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Session Initiation Protocol
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). The protocol defines the specific format of messages exchanged and the sequence of communications for cooperation of the participants. SIP is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP). A call established with SIP may consist of multiple media streams, but no separate streams are required for applications, such as text messaging, that exchange data as payload in the SIP message. SIP works in conjunction with several other protocols that specify and carry the session media. Most commonly, media type and parameter negotiation and media setup are performed with the Session Description ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Voice Over Internet Protocol
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 (voice, fax, SMS, voice-messaging) over the Internet, rather than via the public switched telephone network (PSTN), also known as plain old telephone service (POTS). Overview The steps and principles involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a circuit-switched network, the digital information is packetized and transmission occurs as IP packets over a packet-switched network. They transport media streams using special media delivery protocols t ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
User Agent
In computing, a user agent is any software, acting on behalf of a user, which "retrieves, renders and facilitates end-user interaction with Web content". A user agent is therefore a special kind of software agent. Some prominent examples of user agents are web browsers and email readers. Often, a user agent acts as the client in a client–server system. In some contexts, such as within the Session Initiation Protocol (SIP), the term ''user agent'' refers to both end points of a communications session. User agent identification When a software agent operates in a network protocol, it often identifies itself, its application type, operating system, device model, software vendor, or software revision, by submitting a characteristic identification string to its operating peer. In HTTP, SIP,RFC 3261, ''SIP: Session Initiation Protocol'', IETF, The Internet Society (2002) and NNTP protocols, this identification is transmitted in a header field ''User-Agent''. Bots, such as Web ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Signaling Gateway
A signaling gateway is a network component responsible for transferring signaling messages (i.e. information related to call establishment, billing, location, short messages, address conversion, and other services) between Common Channel Signaling (CCS) nodes that communicate using different protocols and transports. Transport conversion is often from SS7 to IP. A SIGTRAN Signaling Gateway is a network component that performs packet level translation of signaling from common channel signaling (based upon SS7) to SIGTRAN signaling (based upon IP). The concept of the SIGTRAN signaling gateway was introduced in the IETF document: RFC 2719: Architectural Framework for Signaling Transport. A signaling gateway can be implemented as an embedded component of some other network element, or can be provided as a stand-alone network element. For example: a signaling gateway is often part of a softswitch in modern VoIP deployments. The signaling gateway function can also be included withi ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Session Border Controller
A session border controller (SBC) is a network element deployed to protect SIP based voice over Internet Protocol (VoIP) networks. Early deployments of SBCs were focused on the borders between two service provider networks in a peering environment. This role has now expanded to include significant deployments between a service provider's access network and a backbone network to provide service to residential and/or enterprise customers. The term "session" refers to a communication between two or more parties – in the context of telephony, this would be a call. Each call consists of one or more call signaling message exchanges that control the call, and one or more call media streams which carry the call's audio, video, or other data along with information of call statistics and quality. Together, these streams make up a session. It is the job of a session border controller to exert influence over the data flows of sessions. The term "border" refers to a point of demarcation b ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Real-time Transport Protocol
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. RTP typically runs over User Datagram Protocol (UDP). RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is one of the technical foundations of Voice over IP and in this context is often used in conjunction with a signaling protocol such as the Session Initiation Protocol (SIP) which establishes connections across the network. RTP was developed by the Audio-Video Transport Working Group of the Internet Engineering Task Force (IETF) and first published in 1996 a ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Voice Over IP
Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of speech, 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 (voice, fax, Short Message Service, SMS, voice-messaging) over the Internet, rather than via the public switched telephone network (PSTN), also known as plain old telephone service (POTS). Overview The steps and principles involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a circuit-switched network, the digital information is packetized and transmission occurs as IP packets over a packet-switched network. They transport media streams using spec ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Telephony
Telephony ( ) is the field of technology involving the development, application, and deployment of telecommunication services for the purpose of electronic transmission of voice, fax, or data, between distant parties. The history of telephony is intimately linked to the invention and development of the telephone. Telephony is commonly referred to as the construction or operation of telephones and telephonic systems and as a system of telecommunications in which telephonic equipment is employed in the transmission of speech or other sound between points, with or without the use of wires. The term is also used frequently to refer to computer hardware, software, and computer network systems, that perform functions traditionally performed by telephone equipment. In this context the technology is specifically referred to as Internet telephony, or voice over Internet Protocol (VoIP). Overview The first telephones were connected directly in pairs. Each user had a separate telephone wired ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
IP Multimedia Subsystem
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. IMS was originally designed by the wireless standards body 3rd Generation Partnership Project (3GPP), as a part of the vision for evolving mobile networks beyond GSM. Its original formulation (3GPP Rel-5) represented an approach for delivering Internet services over GPRS. This vision was later updated by 3GPP, 3GPP2 and ETSI TISPAN by requiring support of networks other than GPRS, such as Wireless LAN, CDMA2000 and fixed lines. IM ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
