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Cisco ATA 190 Analog Telephone Adaptor Administration Guide for SIP (Version 1.0)
OL-31821-01
Chapter 1 Cisco ATA 190 Analog Telephone Adapter Overview
Session Initiation Protocol Overview
Session Initiation Protocol Overview
Session Initiation Protocol (SIP) is the Internet Engineering Task Force (IETF) standard for real-time
calls and conferencing over Internet Protocol (IP). SIP is an ASCII-based, application-layer control
protocol (defined in RFC3261) that can be used to establish, maintain, and terminate multimedia
sessions or calls between two or more endpoints.
Like other Voice over IP (VoIP) protocols, SIP is designed to address the functions of signaling and
session management within a packet telephony network. Signaling allows call information to be carried
across network boundaries. Session management provides the ability to control the attributes of an
end-to-end call.
Note SIP for the ATA 190 is compliant with RFC2543.
This section contains these topics:
• SIP Capabilities, page 1-2
• Components of SIP, page 1-2
SIP Capabilities
SIP provides these capabilities:
• Determines the availability of the target endpoint. If a call cannot be completed because the target
endpoint is unavailable, SIP determines whether the called party is already on the phone or did not
answer in the allotted number of rings. SIP then returns a message indicating why the target endpoint
was unavailable.
• Determines the location of the target endpoint. SIP supports address resolution, name mapping, and
call redirection.
• Determines the media capabilities of the target endpoint. Using the Session Description Protocol
(SDP), SIP determines the lowest level of common services between endpoints. Conferences are
established using only the media capabilities that are supported by all endpoints.
• Establishes a session between the originating and target endpoint. If the call can be completed, SIP
establishes a session between the endpoints. SIP also supports mid-call changes, such as adding
another endpoint to the conference or changing the media characteristic or codec.
• Handles the transfer and termination of calls. SIP supports the transfer of calls from one endpoint
to another. During a call transfer, SIP establishes a session between the transferee and a new
endpoint (specified by the transferring party) and terminates the session between the transferee and
the transferring party. At the end of a call, SIP terminates the sessions between all parties.
Conferences can consist of two or more users and can be established using multicast or multiple
unicast sessions.
Components of SIP
SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). A user agent can
function in one of these roles:
• User agent client (UAC)—A client application that initiates the SIP request.
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