Describe the characteristics of SIP and explain when to use it

Exam: 642-437 - Cisco Voice over IP (CVOICE)

SIP stands for session initiation protocol. It is a voice servicing protocol which interconnects different VoIP systems and networks. Sip has been prepared as a substitute to H.323 by IEFT (internet engineering task force). The execution of SIP is quiet simplified. It is basically used with gateways and proxy servers. SIP has been prepared to fulfill the signaling functions and session management.

The basic feature on which sip operates is session invitations that are based on HTTP. Sip invites participants into established sessions. Sip enables proxy servers for locating session participants and other functions. SIP is a swift. It is device which creates, modifies and terminates sessions independently without any dependency on the type of session. SIP uses ASCII text based messages for communication as compared to H.323. It analysis the signaling content and allows troubleshooting easily.

Signaling and Deployment

There are five methods that are used for establishing and terminating multimedia communications. Establishing and terminating multimedia communications produces the appended results as shown below in the functionalities that SIP can provide:

  • SIP aids in name mapping, address resolution and redirecting calls.
  • Determination of media capabilities of the target endpoint: SIP provides determination of the least level of common services that exist between the endpoints. Multimedia capabilities that are supported by end points are used for establishing conferences.
  • Determination of availability of the target endpoint: In case due to unavailability of the target endpoint, SIP undertakes determination if the party to whom the call has been made is already connected to a call or has not answered the call within the specified number of rings. Once the reason for non availability is determined a message is returned reporting the same.
  • Establishing a session between the originating and target endpoints: In case the call is capable of being completed, SIP establishes a session between the originating and the target points. It is also possible to undertake changes in the middle of a call. For example an end point can be introduced during a call.
  • Managing the transfer and termination of calls: SIP allows transfer of calls between endpoints.

The levels in a session are called user agents. There can be two types of roles for UA's

  • User agent client-this refers to the start of a SIP request
  • User agent server- it is contacting a user on receiving a SIP invitation and replying back to the origin.

A user agent can play any one of the above said role at a time. It cannot function both ways at a given time. Basically UAC starts a session and UAS ends the session.

The physical components are grouped as appended underneath

  1. Clients-
    • Phone- IP telephone acts as a UAS or UAC on session basis
    • Gateway- it acts as a UAS or UAC and provided call control support

  2. Servers- Registrar, proxy, redirect, and location

    SIP servers

    • Registrar Server- Also located near a location server, the basic purpose is to register the current location.
    • Proxy server- it is a mid way component which receives a SIP request and transfers it to the next SIP Server in the network. The next server can be another proxy server or a UAS.
    • Redirect Server- As the name suggests, it redirects the message to a client about the next hop.
    • Location Server- it is the implementation of mechanism to resolve addresses.
SIP Components

SIP Components


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