Cisco 7960G Administration Manual page 22

Session Initiation Protocol Overview
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Session Initiation Protocol Overview
Session Initiation Protocol (SIP) is the Internet Engineering Task Force (IETF) standard for multimedia
conferencing over IP. SIP is an ASCII-based, application-layer control protocol (defined in RFC 3261)
that can be used to establish, maintain, and terminate calls between two or more endpoints.
Like other 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.
SIP Capabilities
SIP provides the capabilities to do the following:
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Cisco Unified IP Phone 7960G and 7940G Administration Guide for Release 8.0 (SIP)
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Access port (10 and 100 PC) RJ-45 to connect a network device, such as a computer, to the phone
supporting from 10- to 100- Mbps half- or full-duplex Ethernet connections to external devices.
You can use either Category 3 or Category 5 cabling for 10-Mpbs connections, but use Category 5
for 100-Mbps connections. To avoid collisions, use full-duplex mode. You must use a
straight-through cable on this port.
Handset port for connecting a handset.
Headset port for connecting a headset. Enables the headset. The phone supports a four- or six-wire
headset jack. The volume and mute controls also adjust volume to the earpiece and mute the speech
path of the headset. The headset activation key is located on the front of the Cisco Unified IP Phone
7960G and 7940G.
The phone supports the following Plantronics four- or six-wire headsets: Tristar Monaural, Encore
Monaural H91, and Encore Binaural H101.
When a headset is used, an amplifier is not required. However, a coil cord is required to connect
the headset to the headset port on the back of your Cisco Unified IP Phone 7960G and 7940G. For
information on ordering compatible headsets and coil cords for the Cisco Unified IP Phone 7960G
and 7940G, go to http://cisco.plantronics.com
Determine the location of the target endpoint—SIP supports address resolution, name mapping, and
call redirection.
Determine the media capabilities of the target endpoint—Using Session Description Protocol
(SDP), SIP determines the "lowest level" of common services between the endpoints. Conferences
are established using only the media capabilities that can be supported by all endpoints.
Determine 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. It then returns a message that indicates why the target
endpoint was unavailable.
Establish a session between the originating and target endpoint—If the call can be completed, SIP
establishes a session between the endpoints. SIP also supports midcall changes, such as the addition
of another endpoint to the conference or the changing of a media characteristic or codec.
Handle the transfer and termination of calls—SIP supports the transfer of calls from one endpoint
to another. During a call transfer, SIP simply 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.
Chapter 1
or http://www.vxicorp.com/cisco.
Product Overview
OL-7890-01