SIP Protocol Overview

Advanced Telephony on the Internet

Session-Initiation Protocol (hereafter referred to as SIP) is signaling protocol for Internet conferencing, telephony, presence, event notification, and instant messaging.

Dialogic not only SIP-enables its platform products but stays current as the protocol evolves.

SIP is typically required in a softswitch-controlled, converged network. In these converged networks, media gateways handle circuit/packet conversions (usually between IP and voice) and require media services such as tones, prompts, conferencing, and announcements.

Dialogic introduced SIP software to meet the demand for IP in converged services networks. This feature allows the CSP to act as an IP service node, providing application services and media resources to a softswitch or proxy server. The softswitch or SIP proxy server uses SIP to hand off a call requiring call treatment via partner-developed applications resident on a CSP. This provides services in Real-Time Protocol (RTP) streams to media gateways.

The SIP software is embedded in the CSP Matrix Series 3 Card, and interacts with host applications the same way that other Layer 3 circuit-based protocols do, such as Integrated Services Digital Network (ISDN) and SS7 Integrated Services Digital Network User Part (SS7 ISUP).

What SIP Allows

SIP allows the CSP to act as an IP Service Node, providing application services and media resources to a softswitch or SIP proxy server. A softswitch or SIP proxy server uses SIP to hand off a call requiring call treatment via partner-developed applications resident on a host.

Services Provided by SIP

SIP is part of the Internet Engineering Task Force (IETF) standards process (RFC 2543 BIS) and is modeled upon other Internet protocols, such as SMTP (Simple Mail Transfer Protocol) and HTTP (Hypertext Transfer Protocol.) It is used to establish, change, and tear down calls between one or more users in an IP-based network.


SIP Protocol Elements and Architecture

SIP is comprised of the following six elements in its architecture:

User Agent Client (UAC)

User Agent Server (UAS)

SIP Terminal

Proxy Server

Redirect Server

Location Server

These elements are grouped as follows:

User Agent

The User Agent is effectively the end system component for the call and has:

The User Agent Client (UAC) as the client

The User Agent Server (UAS) as the server

The client element initiates the calls and the server element answers the calls. This allows peer-to-peer calls to be made using a client-server protocol. A SIP terminal is a device, such as a SIP phone, that supports two-way, real time communications in a SIP network.

The CSP acts as the User Agent and derivations of a basic User Agent.

The SIP Network Server

The SIP Network Server handles the signaling associated with multiple calls. There are three types of SIP Network Server elements:

Proxy Server

Redirect Server

Location Server

The Proxy Server receives a SIP request, then passes the request along to one or more clients or next-hop servers. The Redirect Server accepts the SIP request, determines the new address, and returns the new IP address to the SIP client. The Location Server under SIP provides the current IP address of clients to the Redirect Server and to the Proxy Server on the network.

The following diagram shows a SIP based network architecture with the high-level functionality provided by the CSP. The network can be 3G Wireless or Wireline.

Figure 5-1 SIP Based Network Architecture



SIP must provide or enable the following functions:

Name Translation and User Location

This function ensures:

The call reaches the called party wherever they are located.

Descriptive information is mapped to location information.

The details of the nature of the session (call) are supported.

Feature Negotiation

This function allows the group involved in a call (this may be a multi-party call) to agree on the supported features, recognizing that not all the parties can support the same level of features. For example, video may or may not be supported. SIP supports all Multipurpose Internet Mail Extensions (MIME). There is plenty of scope for negotiation.

Call Participant Management

During a call, a participant can bring other users into the call or cancel connections to current users. In addition, users could be transferred or placed on hold.

Call Feature Changes

A participant is able to change the call characteristics during the course of the call. For example, a call may have been set up as voice-only, but in the course of the call, the participant may need to enable a audio or fax function. A third party joining a call may need different features enabled in order to participate in the call.