Cisco 7940G Yönetici El Kitabı

IP Telefon Cisco 7940G için çevrimiçi göz atın veya pdf Yönetici El Kitabı indirin. Cisco 7940G 15 sayfaları. Ios telephony services ip phone
Ayrıca Cisco 7940G için: Kullanıcı Kılavuzu (21 sayfalar), Kullanıcı Kılavuzu (25 sayfalar), Veri Sayfası (7 sayfalar), Manuel (35 sayfalar), Hızlı Referans (2 sayfalar), Kullanıcı Kılavuzu (15 sayfalar), Hızlı Referans Kılavuzu (2 sayfalar), Kullanıcı Kılavuzu (20 sayfalar), Telefon Kılavuzu (17 sayfalar), Kullanıcı Kılavuzu (16 sayfalar), Referans Kılavuzu (24 sayfalar), Yönetim Kılavuzu (29 sayfalar), Manuel (21 sayfalar), Kurulum Kılavuzu (12 sayfalar), Kullanıcı Kılavuzu (10 sayfalar)

Product Overview

This chapter contains the following information about the Cisco SIP IP phone:

What Is Session Initiation Protocol?, page 1-1

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What Is the Cisco SIP IP Phone?, page 1-3

Prerequisites, page 1-10

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Cisco SIP IP Phone Connections, page 1-10

The Cisco SIP IP Phone with a Catalyst Switch, page 1-13

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What Is Session Initiation Protocol?

Session Initiation Protocol (SIP) is the Internet Engineering Task Force's (IETF's) standard for

multimedia conferencing over IP. SIP is an ASCII-based, application-layer control protocol (defined in

RFC 2543) 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 provides the capabilities to:

Determine the location of the target endpoint—SIP supports address resolution, name mapping, and

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call redirection.

Determine the media capabilities of the target endpoint—Via Session Description Protocol (SDP),

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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

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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 indicating why the target endpoint

was unavailable.

Establish a session between the originating and target endpoint—If the call can be completed, SIP

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establishes a session between the endpoints. SIP also supports mid-call changes, such as the addition

of another endpoint to the conference or the changing of a media characteristic or codec.

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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.

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C H A P T E R

Cisco SIP IP Phone Administrator Guide Version 3.0

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