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Answer-seizure ratio

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Jump to: navigation, search The answer-seizure ratio (ASR) is a measurement of network quality[1] and call success rates in telecommunications. It is the percentage of answered telephone calls with respect to the total call volume.

Definition[edit source | editbeta]
In telecommunication an attempted call is termed a seizure. The answer-seizure ratio is defined as 100 times the ratio of answered calls, i.e. the number of seizures resulting in an answer signal, divided by the total number of seizures:

Busy signals and other call rejections by the telephone network count as call failures. However, the inclusion in the ASR accounting of some failed calls varies in practical applications. This makes the ASR highly dependent on end-user action. Low answer-seizure ratios may be caused by far-end switch congestion, not answering by called parties and busy destination circuits.

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session border controller (SBC)
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A session border controller (SBC) is a device or application that governs the manner in which calls, also called sessions, are initiated, conducted and terminated in a VoIP (Voice over Internet Protocol) network. An SBC can be placed in the communication path between any two parties engaged in a VoIP session. A common location for a stand-alone SBC is a connection point, called a border, between a private local area network (LAN) and the Internet. An SBC can exist as a single, independent hardware unit containing all of the resources necessary for call signalingand call control. Alternatively, the signaling and control functions can be divided between the communicating systems. An SBC can facilitate VoIP sessions between phone sets or proprietary networks that use different signaling protocols. In addition, an SBC can include call filtering, bandwidth use management, firewalls and anti-malwareprograms to minimize abuse and enhance security. An SBC can act as a router and conceal the internal topology of a private network from the external environment. However, regulations demand that SBCs allow interception and monitoring of communications for law enforcement purposes. Emergency calls are given top priority so they can be completed under all network traffic conditions.

The next-generation network (NGN) is 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 (voice, data, and all sorts of media such as video) by encapsulating these into 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 toward NGN. Jump to: navigation, search

Media Gateways are used for transcoding media between PSTN and IP networks Media Gateway Controller - Coordinates setup, handing and termination of media flows at the media gateway. Signalling Gateway - SS7-IP interface, coordinates SS7 view of IP elements and Ip view of SS7 elements. Media Gateway - Terminates PSTN lines and packetizes media streams for IP transport.

A media gateway is a translation device or service that converts digital media streams between disparate telecommunications networks such as PSTN, SS7, Next Generation Networks (2G, 2.5G and 3G radio access networks) or PBX. Media gateways enable multimedia communications across Next Generation Networks over multiple transport protocols such as Asynchronous Transfer Mode (ATM) and Internet Protocol (IP). Because the media gateway connects different types of networks, one of its main functions is to convert between different transmission and coding techniques (see also Transcode). Media streaming functions such as echo cancellation, DTMF, and tone sender are also located in the media gateway. Media gateways are often controlled by a separate Media Gateway Controller which provides the call control and signaling functionality. Communication between media gateways and Call Agents is achieved by means of protocols such as MGCP or Megaco (H.248) or Session Initiation Protocol (SIP). Modern media gateways used with SIP are often stand-alone units with their own call and signaling control integrated and can function as independent, intelligent SIP end-points. Voice over Internet Protocol (VoIP) media gateways perform the conversion between TDM voice to a media streaming protocol (usually Real-time Transport Protocol, RTP), as well as a signaling protocol used in the VoIP system. Mobile access Media Gateways connect the radio access networks of a public land mobile network PLMN to a Next Generation Core network. 3GPP standards define the functionality of CS-MGW and IMS-MGW for UTRAN and GERAN based PLMNs.

Jump to: navigation, search A Signal Transfer Point (STP) is a router that relays SS7 messages between signaling endpoints (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. These 'routers' are connected just by signaling links; they don't have users attached (where a user could be a mobile station (MS), a PSTN user in case of a public terrestrial network, or a piece of terminal equipment at the end of an ISDN B channel). SEPs send signaling messages to other SEPs, but the messages are normally routed via the SEP's adjacent STPs. An STP's main function is to identify the best path for two SEPs to communicate. A typical application would be for two SEPs to agree on the use of a shared data path (e.g., using ISUP to initiate a voice call between a user on one SEP and a user on the second SEP). In this way, STPs route signaling messages (for starting, maintaining or finishing any kind of calls originated by the SEPs' attached users) while avoiding disabled intermediary STPs. A signaling message typically never goes directly from a given SEP to the destination SEP: the message would normally have to pass through the initiating SEP's adjacent STP so that it can be routed to the destination SEP. In some applications, however, SEPs might be directly connected with signaling links; this would typically be done to enhance robustness or performance between two critical SEPs. Such mesh network configurations are also common in Europe, where STPs have not found widespread deployment. In some cases, signaling messages can be originated by the STP to learn about the state of the signaling network Some examples include:
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an STP may send route set test messages to probe the availability of a particular SEP; it may send low-level MTP messages to an adjacent signaling point to check the Bit Error Rate (BER) on a particular signaling link; or it may let other adjacent signaling points know that it is going to out of service; in this way, the adjacent signaling points will try to avoid this OOS STP.

A given piece of equipment can implement both SEP and STP functionality. This is commonly done in some SSPs. This is also seen in Signaling Gateways that also have Application Server (AS) functionality as defined by the IETF. Some UMTS number portability solutions are implemented in STPs. In UMTS, the STP provides Global Title Translation (GTT), which may be used to route queries from a gateway MSC (GMSC) to the HLR. Note that for every call to an MS, the call is first routed to the MS's Gateway MSC.