• Document: Overview of SIP. Information About SIP. SIP Capabilities. This chapter provides an overview of the Session Initiation Protocol (SIP).
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Overview of SIP This chapter provides an overview of the Session Initiation Protocol (SIP). • Information About SIP, page 1 • How SIP Works, page 4 • How SIP Works with a Proxy Server, page 5 • How SIP Works with a Redirect Server, page 7 • SIP Call Flows, page 9 • Additional References, page 28 Information About SIP Session Initiation Protocol (SIP) is an ASCII-based, application-layer control protocol that can be used to establish, maintain, and terminate calls between two or more endpoints. SIP is an alternative protocol developed by the Internet Engineering Task Force (IETF) for multimedia conferencing over IP. SIP features are compliant with IETF RFC 2543, SIP: Session Initiation Protocol, published in March 1999. The Cisco SIP implementation enables supported Cisco platforms to signal the setup of voice and multimedia calls over IP networks. 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 following capabilities: • Determines the location of the target endpoint--SIP supports address resolution, name mapping, and call redirection. • Determines the media capabilities of the target endpoint--SIP determines the lowest level of common services between the endpoints through Session Description Protocol (SDP). Conferences are established using only the media capabilities that can be supported by all endpoints. SIP Configuration Guide, Cisco IOS Release 15M&T 1 Overview of SIP SIP Components • 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 connected to a call already or did not answer in the allotted number of rings. SIP then returns a message indicating why the target endpoint was unavailable. • Establishes a session between the originating and target endpoints--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. • Handles 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 among all parties. Note The term “conference” describes an established session (or call) between two or more endpoints. Conferences consist of two or more users and can be established using multicast or multiple unicast sessions. SIP Components SIP is a peer-to-peer protocol. The peers in a session are called user agents (UAs). A UA can function in one of the following roles: • User-agent client (UAC)--A client application that initiates the SIP request. • User-agent server (UAS)--A server application that contacts the user when a SIP request is received and that returns a response on behalf of the user. Typically, a SIP endpoint is capable of functioning as both a UAC and a UAS, but functions only as one or the other per transaction. Whether the endpoint functions as a UAC or a UAS depends on the user agent that initiated the request. From an architectural standpoint, the physical components of a SIP network can be grouped into two categories: clients (endpoints) and servers. The figure below illustrates the architecture of a SIP network. SIP Configuration G

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