Table Of Contents
AMR-NB—Adaptive Multi Rate codec - Narrow Band.
Allow header—Lists the set of methods supported by the UA generating the message.
bind — In SIP, configuring the source address for signaling and media packets to the IP address of a specific interface.
call—In SIP, a call consists of all participants in a conference invited by a common source. A SIP call is identified by a globally unique call identifier. A point-to-point IP telephony conversation maps into a single SIP call.
call leg—A logical connection between the router and another endpoint.
Content-Type header—Specifies the media type of the message body.
CSeq header—Serves as a way to identify and order transactions. It consists of a sequence number and a method. It uniquely identifies transactions and differentiates between new requests and request retransmissions.
delta—An incremental value. In this case, the delta is the difference between the current time and the time when the response occurred. dial peer—An addressable call endpoint.
dial peer—An addressable call endpoint.
DNS—Domain Name System. Used to translate H.323 IDs, URLs, or e-mail IDs to IP addresses. DNS is also used to assist in locating remote gatekeepers and to reverse-map raw IP addresses to host names of administrative domains.
DNS SRV—Domain Name System Server. Used to locate servers for a given service.
DSP—Digital Signal Processor.
DTMF—dual-tone multifrequency. Use of two simultaneous voice-band tones for dialing (such as touch-tone).
EFXS—IP phone virtual voice ports.
FQDN—fully qualified domain name. Complete domain name including the host portion; for example, serverA.companyA.com.
FXS—analog telephone voice ports.
gateway—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.
H.323—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.
iLBC—internet Low Bitrate Codec.
INVITE—A SIP message that initiates a SIP session. It indicates that a user is invited to participate, provides a session description, indicates the type of media, and provides insight regarding the capabilities of the called and calling parties.
IP— Internet Protocol. A connectionless protocol that operates at the network layer (Layer 3) of the OSI model. IP provides features for addressing, type-of-service specification, fragmentation and reassemble, and security. Defined in RFC 791. This protocol works with TCP and is usually identified as TCP/IP. See TCP/IP.
ISDN—Integrated Services Digital Network.
Minimum Timer—Configured minimum value for session interval accepted by SIP elements (proxy, UAC, UAS). This value helps minimize the processing load from numerous INVITE requests.
Min-SE—Minimum Session Expiration. The minimum value for session expiration.
multicast—A process of transmitting 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.
originator—User agent that initiates the transfer or Refer request with the recipient.
PDU—protocol data units. Used by bridges to transfer connectivity information.
PER—Packed Encoding Rule.
proxy—A SIP UAC or UAS that forwards requests and responses on behalf of another SIP UAC or UAS.
proxy server—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.
recipient—User agent that receives the Refer request from the originator and is transferred to the final recipient.
redirect server—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 or accept calls.
re-INVITE—An INVITE request sent during an active call leg.
Request URI—Request Uniform Resource Identifier. It can be a SIP or general URL and indicates the user or service to which the request is being addressed.
RFC—Request For Comments.
RTP—Real-Time Transport Protocol (RFC 1889)
SCCP—Skinny Client Control Protocol.
SDP—Session Description Protocol. Messages containing capabilities information that are exchanged between gateways.
session—A SIP session is a set of multimedia senders and receivers and the data streams flowing between the senders and receivers. A SIP multimedia conference is an example of a session. The called party can be invited several times by different calls to the same session.
session expiration—The time at which an element considers the call timed out if no successful INVITE transaction occurs first.
session interval—The largest amount of time that can occur between INVITE requests in a call before a call is timed out. The session interval is conveyed in the Session-Expires header. The UAS obtains this value from the Session-Expires header of a 2xx INVITE response that it sends. Proxies and UACs determine this value from the Session-Expires header in a 2xx INVITE response they receive.
SIP—Session Initiation Protocol. An application-layer protocol originally developed by the Multiparty Multimedia Session Control (MMUSIC) working group of the Internet Engineering Task Force (IETF). Their goal was to equip platforms to signal the setup of voice and multimedia calls over IP networks. SIP features are compliant with IETF RFC 2543, published in March 1999.
SIP URL—Session Initiation Protocol Uniform Resource Locator. Used in SIP messages to indicate the originator, recipient, and destination of the SIP request. Takes the basic form of user@host, where user is a name or telephone number, and host is a domain name or network address.
SPI—service provider interface.
socket listener— Software provided by a socket client to receives datagrams addressed to the socket.
stateful proxy—A proxy in keepalive mode that remembers incoming and outgoing requests.
TCP—Transmission Control Protocol. Connection-oriented transport layer protocol that provides reliable full-duplex data transmissions. TCP is part of the TCP/IP protocol stack. See also TCP/IP and IP.
UA—user agent. A combination of UAS and UAC that initiates and receives calls. See UAS and UAC.
UAC—user agent client. A client application that initiates a SIP request.
UAS—user agent server. A server application that contacts the user when a SIP request is received and then returns a response on behalf of the user. The response accepts, rejects, or redirects the request.
UDP— User Datagram Protocol. Connectionless transport layer protocol in the TCP/IP protocol stack. UDP is a simple protocol that exchanges datagrams without acknowledgments or guaranteed delivery, requiring that error processing and retransmission be handled by other protocols. UDP is defined in RFC-768.
URI—Uniform Resource Identifier. Takes a form similar to an e-mail address. It indicates the user's SIP identity and is used for redirection of SIP messages.
URL—Universal Resource Locator. Standard address of any resource on the Internet that is part of the World Wide Web (WWW).
User Agent—A combination of UAS and UAC that initiates and receives calls. See UAS and UAC.
VFC—Voice Feature Card.
VoIP—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 approach (for example, H.323) to IP voice traffic.