Authentication, authorization, and accounting. AAA is a suite of network security services that provides the primary framework through which access control can be set up on your Cisco router or access server.
Automatic number identification.
Basic telephony extensible markup language
Call control applications programming interface.
Customer premises equipment. Terminating equipment, such as terminals, telephones, and modems, supplied by the telephone company, installed at the customer sites, and connected to the telephone company network.
Call switching module.
An addressable call endpoint. In Voice over IP (VoIP), there are two types of dial peers: POTS and VoIP.
Domain Name System. Used to address translation to convert H.323 IDs, URLs, or e-mail IDs to IP addresses. DNS is also used to assist in the locating remote gatekeepers and to reverse-map raw IP addresses to host names of administrative domains.
Dialed number identification service (the called number)
Digital signal processor.
Dual tone multifrequency.
The international public telecommunications numbering plan. A standard set by ITU-T which addresses telephone numbers.
Ear and mouth RBS signaling.
A SIP terminal or gateway. An endpoint can call and be called. It generates and/or terminates the information stream.
A gateway allows SIP or H.323 terminals to communicate with terminals configured to other protocols by converting protocols. A gateway is the point where a circuit-switched call is encoded and repackaged into IP packets.
An International Telecommunication Union (ITU-T) standard that describes packet-based video, audio, and data conferencing. H.323 is an umbrella standard that describes the architecture of the conferencing system and refers to a set of other standards (H.245, H.225.0, and Q.931) to describe its actual protocol.
Registration, admission, and status. The RAS signaling function performs registration, admissions, bandwidth changes, status and disengage procedures between the VoIP gateway and the gatekeeper.
Interactive voice response. When someone dials in, IVR responds with a prompt to get a personal identification number (PIN), and so on.
Local exchange carrier.
A SIP redirect or proxy server uses a a location service to get information about a caller's locations. Location services are offered by location servers.
Multifrequency tones are made of six frequencies that provide 15 two frequency combinations for indication digits 0-9 and KP/ST signals.
A process of transmitting protocol data units (PDUs) from one source to many destinations. The actual mechanism (that is, IP multicast, multi-unicast, and so forth) for this process might be different for LAN technologies.
A process of transferring PDUs where an endpoint sends more than one copy of a media stream to different endpoints. This can be necessary in networks that do not support multicast.
A H.323 entity that uses RAS to communicate with the gatekeeper; for example, an endpoint such as a terminal, proxy, or gateway.
Protocol data units used by bridges to transfer connectivity information.
Plain old telephone service. Basic telephone service supplying standard single line telephones, telephone lines, and access to the PSTN.
An intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. A proxy interprets, and, if necessary, rewrites a request message before forwarding it.
Public switched telephone network. PSTN refers to the local telephone company.
A redirect server is a server that accepts a SIP request, maps the address into zero or more new addresses and returns these addresses to the client. It does not initiate its own SIP request nor accept calls.
A registrar is a server that accepts REGISTER requests. A registrar is typically colocated with a proxy or redirect server and
may offer location services.
Registration, admission, and status protocol. This is the protocol that is used between endpoints and the gatekeeper to perform management functions.
Session Initiation Protocol. This is a protocol developed by the IETF MMUSIC Working Group as an alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March 1999.
SIP equips platforms to signal the setup of voice and multimedia calls over IP networks.
Service provider interface.
Time-division multiplexing. Technique in which information from multiple channels can be allocated bandwidth on a single wire, based on preassigned time slots. Bandwidth is allocated to each channel regardless of whether the station has data to transmit.
User agent client. A user agent client is a client application that initiates the SIP request.
User agent server (or user agent). A user agent server is a server application that contacts the user when a SIP request is received, then returns a response on behalf of the user. The response accepts, rejects, or redirects the request.
Voice over IP. The ability to carry normal telephone-style voice over an IP-based Internet with POTs-like functionality, reliability, and voice quality. VoIP is a blanket term, that generally refers to the Cisco standards-based (for example H.323) approach to IP voice traffic.