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.
The Internet Protocol (IP) is the principal communications protocol in the Internet protocol suite for relaying datagrams across network boundaries. Its routing function enables internetworking, and essentially establishes the Internet.
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.
SIGTRAN is the name, derived from signaling transport, of the former Internet Task Force (I) working group that produced specifications for a family of protocols that provide reliable datagram service and user layer adaptations for Signaling System and ISDN communications protocols. The SIGTRAN protocols are an extension of the SS7 protocol family. It supports the same application and call management paradigms as SS7 but uses an Internet Protocol (IP) transport called Stream Control Transmission Protocol (SCTP), instead of TCP or UDP. Indeed, the most significant protocol defined by the SIGTRAN group is SCTP, which is used to carry PSTN signaling over IP.
The Internet Engineering Task Force (IETF) is an open standards organization, which develops and promotes voluntary Internet standards, in particular the standards that comprise the Internet protocol suite (TCP/IP). It has no formal membership or membership requirements. All participants and managers are volunteers, though their work is usually funded by their employers or sponsors.
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 within the larger operational domain of a Signal Transfer Point (STP).
A softswitch is a central device in a telecommunications network which connects telephone calls from one phone line to another, across a telecommunication network or the public Internet, entirely by means of software running on a general-purpose computer system. Most landline calls are routed by purpose-built electronic hardware; however, soft switches using general purpose servers and VoIP technology are becoming more popular.
A Signal Transfer Point (STP) is a router that relays SS7 messages between signaling end-points (SEPs) and other signaling transfer points (STPs). Typical SEPs include service switching points (SSPs) and service control points (SCPs). The STP is connected to adjacent SEPs and STPs via signaling links. Based on the address fields of the SS7 messages, the STP routes the messages to the appropriate outgoing signaling link. Edge STPs can also route based upon message body content using deep packet inspection techniques, and can provide address translations and screen content to limit the transfer of messages with dubious content or sent from unreliable sources. To meet stringent reliability requirements, STPs are typically provisioned in mated pairs.
Protocol conversion gateways can also convert from one network operational paradigm to another – for example, SIP to ISUP for call control, SIP to TCAP for address translation, or SIP to MAP for location or presence.
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).
The ISDN User Part or ISUP is part of Signaling System No. 7 (SS7), which is used to set up telephone calls in the public switched telephone network (PSTN). It is specified by the ITU-T as part of the Q.76x series.
Transaction Capabilities Application Part, from ITU-T recommendations Q.771-Q.775 or ANSI T1.114 is a protocol for Signalling System 7 networks. Its primary purpose is to facilitate multiple concurrent dialogs between the same sub-systems on the same machines, using Transaction IDs to differentiate these, similar to the way TCP ports facilitate multiplexing connections between the same IP addresses on the Internet.
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Voice over Internet Protocol (VoIP), also called IP telephony, is a methodology 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 public Internet, rather than via the public switched telephone network (PSTN).
Signaling System No. 7 (SS7) is a set of telephony signaling protocols developed in 1975, which is used to set up and tear down telephone calls in most parts of the world-wide public switched telephone network (PSTN). The protocol also performs number translation, local number portability, prepaid billing, Short Message Service (SMS), and other services.
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).
The Message Transfer Part (MTP) is part of the Signaling System 7 (SS7) used for communication in Public Switched Telephone Networks. MTP is responsible for reliable, unduplicated and in-sequence transport of SS7 messages between communication partners.
A global title (GT) is an address used in the SCCP protocol for routing signaling messages on telecommunications networks. In theory, a global title is a unique address which refers to only one destination, though in practice destinations can change over time.
A public-safety answering point (PSAP), sometimes called "public-safety access point", is a call center in Canada and the United States responsible for answering calls to an emergency telephone number for police, firefighting, and ambulance services. Trained telephone operators are also usually responsible for dispatching these emergency services. Most PSAPs are now capable of caller location for landline calls, and many can handle mobile phone locations as well, where the mobile phone company has a handset location system. Some can also use voice broadcasting where outgoing voice mail can be sent to many phone numbers at once, in order to alert people to a local emergency such as a chemical spill.
The IP Multimedia Subsystem or IP Multimedia Core Network Subsystem (IMS) is an 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 to provide such standardization.
A Session Border Controller (SBC) is a network element deployed to protect SIP based Voice over Internet Protocol (VoIP) networks.
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, instead of the traditional public switched telephone network (PSTN).
The Signalling Connection Control Part (SCCP) is a network layer protocol that provides extended routing, flow control, segmentation, connection-orientation, and error correction facilities in Signaling System 7 telecommunications networks. SCCP relies on the services of MTP for basic routing and error detection.
A back-to-back user agent (B2BUA) is a logical network element in Session Initiation Protocol (SIP) applications. SIP is a signaling protocol to manage multimedia Voice over Internet Protocol (VoIP) telephone calls. A back-to-back user agent operates between both end points of a phone call or communications session and divides the communication channel into two call legs and mediates all SIP signaling between both ends of the call, from call 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 context of computer networking, an application-level gateway consists of a security component that augments a firewall or NAT employed in a computer network. It allows customized NAT traversal filters to be plugged into the gateway to support address and port translation for certain application layer "control/data" protocols such as FTP, BitTorrent, SIP, RTSP, file transfer in IM applications, etc. In order for these protocols to work through NAT or a firewall, either the application has to know about an address/port number combination that allows incoming packets, or the NAT has to monitor the control traffic and open up port mappings dynamically as required. Legitimate application data can thus be passed through the security checks of the firewall or NAT that would have otherwise restricted the traffic for not meeting its limited filter criteria.
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.
HP OpenCall is an extensive portfolio of network and telephony products offered by the HP Software Division of technology company HP. It is most commonly described as a suite of software and hardware applications which allow implementation of common telecom operator services such as voicemail, sms, prepaid, billing, hlr, etc. It implements industry standard telecom protocols and standards such as SS7, ISUP, TCAP, SIP, MRCP, RTSP, and VoiceXML.
Edge STPs are networking hardware devices embedded with software that performs routing, signaling, firewall, and packet conversion functions. Their primary purpose is to unify networks that use various transports and signaling protocols – such as SS7, SIP, SIGTRAN, TDM, IP, etc. – into cohesive service environments. Unified environments are simpler for telecommunications companies to manage, and also enable them to cost-effectively transition to next-generation networks based on the Internet Protocol (IP).
The Media Gateway Control Protocol (MGCP) is a signaling and call control communications protocol used in voice over IP (VoIP) telecommunication systems. It implements the media gateway control protocol architecture for controlling media gateways on Internet Protocol (IP) networks connected to the public switched telephone network (PSTN). The protocol is a successor to the Simple Gateway Control Protocol (SGCP), which was developed by Bellcore and Cisco, and the Internet Protocol Device Control (IPDC).
The Stream Control Transmission Protocol (SCTP) is a computer networking communications protocol which operates at the transport layer and serves a role similar to the popular protocols TCP and UDP. It is standardized by IETF in RFC 4960.
WebRTC Gateway connects between WebRTC and an established VoIP technology such as SIP. WebRTC is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, and messaging without the need of either internal or external plugins.
