для Cisco Cisco IOS Software Release 12.2(15)ZJ

call—In SIP, a call consists of all participants in a conference who are 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. 

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.

DTMF—dual-tone multifrequency. Tones that are generated when a button on a touch-tone phone is 
pressed. When the tone is generated, it is compressed, transported to the other party, and decompressed.

DTMF relay— DTMF relay provides reliable digit relay between VoIP gateways when a low-bandwidth 
codec is used. DTMF relay provides a standardized means of transporting DTMF tones in Real-Time 
Transport Protocol (RTP) packets and is identified by dynamic payload types in the SDP.

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 information regarding the 
capabilities of the called and calling parties. 

NOTIFY—SIP NOTIFY messages report when certain events occur, such as DTMF events.

proxy—A SIP UAC or UAS that forwards requests and responses on behalf of another SIP UAC or UAS.

PSTN—public switched telephone network. PSTN refers to the local telephone company network.

SCCP—Skinny Client Control Protocol. SCCP is the Cisco standard for real-time calls and conferencing 
over IP. This generalized messaging set allows Cisco IP phones to coexist in an H.323 environment. The 
savings in memory size, processor power, and complexity makes the protocol desirable. 

session—A SIP session includes a set of multimedia senders and receivers and the data streams that flow 
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.

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.

SRST—Survivable Remote Site Telephony.

URI—Uniform Resource Identifier. Takes a form similar to an e-mail address, indicates the user’s SIP 
identity, and is used for redirection of SIP messages.

URL—Uniform Resource Locator. Standard address of any resource on the Internet that is part of the 
World Wide Web (WWW).

VoIP—Voice over IP. The ability to carry normal telephone-style voice over an IP-based network.

Refer to the

 for terms not included in this glossary.