Ata chapter wiring
In addition, the gateway also translates between audio and video codecs and performs call setup and clearing onīoth the LAN side and the switched-circuit network side. This function includes translation between transmission formats and between communications Gateways provide many services, the most common being a translation function between SIP conferencingĮndpoints and other terminal types. Such as directory, authentication, and billable services. These application services provide back-end services SIP servers can interact with other application services, such as Lightweight Directory Access Protocol (LDAP) servers, aĭatabase application, or an extensible markup language (XML) application. The following figure shows the architecture of a SIP network. Whether the endpoint functions as a UAC or a UAS depends on the UA that initiated the request.įrom an architectural standpoint, the physical components of a SIP network can also be grouped into two categories-ClientsĪnd servers.
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Typically, a SIP endpoint is capable of functioning as both a UAC and a UAS, but functions only as one or the other per transaction. User agent server (UAS)-A server application that contacts the user when a SIP request is received and returns a response User agent client (UAC)-A client application that initiates the SIP request. A user agent can function in one of these The peers in a session are called User Agents (UAs). Conferences can consist of two or more users and can be established using multicast or multiple unicast At the end of a call, SIP terminates the sessionsīetween all parties. Terminates the session between the transferee and the transferring party. Transfer, SIP establishes a session between the transferee and a new endpoint (specified by the transferring party). SIP supports the transfer of calls from one endpoint to another. Handles the transfer and termination of calls. SIP also supports midcall changes, such as adding another endpoint to the conference or changing the If the call can be completed, SIP establishes a sessionīetween the endpoints. Conferences are established using only the media capabilities that allĮstablishes a session between the originating and target endpoint. Lowest level of common services between endpoints. Using the Session Description Protocol (SDP), SIP determines the SIP supports address resolution, name mapping, and call redirection.ĭetermines the media capabilities of the target endpoint. SIP then returns a message indicating whyĭetermines the location of the target endpoint. Party is already on the phone or didn't answer in the allotted number of rings. If the target endpoint is unavailable, SIP determines whether the called Session Initiation Protocol (SIP) provides these capabilities:ĭetermines the availability of the target endpoint. SIP for the ATA 191 is compliant with RFC2543. Is used to control the attributes of an end-to-end call. Signaling allows call information to be carried across network boundaries.
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Like other Voice over IP (VoIP) protocols, SIP is designed to address the functions of signaling and session management withinĪ packet telephony network. It is used toĮstablish, maintain, and terminate multimedia sessions or calls between two or more endpoints. SIP is an ASCII-based, application-layer control protocol (defined in RFC3261). Session Initiation Protocol (SIP) is the Internet Engineering Task Force (IETF) standard for real-time calls and conferencing The ATA 191 supports two voice ports, each with an independent phone number. The ATA 191 analog telephone adapter is a telephony-device-to-Ethernet adapter that allows regular analog phones to operate
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Cisco ATA 191 Analog Telephone Adapter Overview