Sip Signaling

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SIP signaling
Contents
This section contains information on the following topics: Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . SIP requests and responses . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Format of a SIP message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .Direct IP Media Paths . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . IP Phone to IP Phone (on separate Call Servers) . . . . . . . . . . . . . . . Call scenarios . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53 55 56 57 59 70

Introduction
The SIP Trunk Gateway offers an industry-standard SIP-based IP Peer solution. A SIP TrunkGateway delivers a SIP interface for interoperability with Nortel SIP products and other industry SIP-based products. The SIP Trunk Gateway is implemented according to SIP standards. The SIP Trunk Gateway can connect two CS 1000 nodes and can also connect CS 1000 systems to other Nortel or third-party SIP-enabled products. This SIP Trunk Gateway interworks with the MCS 5100 system. The SIP trunkingapplication resides on the Signaling Server. The SIP Trunk Gateway provides a direct trunking interface between the CS 1000 systems and a SIP domain. For information, see “SIP Trunk Gateway software — trunk route redundancy” on page 187.

IP Peer Networking

Installation and Configuration

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SIP signaling

Figure 2 shows the CS 1000 SIP Trunk Gateway interworking.
Figure2 CS 1000 SIP Trunk Gateway interworking
Signaling Server (Optionally Redundant) - SIP Gateway (Virtual Trunk) - H.323 Gateway (Virtual Trunk) - IP LIne TPS - web browser

CS 1000

IP Phone

MCS 5100 Call Server WAN/LAN SIP Domain Other systems

IP Phone Voice Gateway Media Card

SIP Phones

CS 1000 system
553-AAA2341

The direct SIP interface provides the following: • • removesthe requirement for a SIP/PRI gateway between the MCS 5100 and the CS 1000 systems improves voice quality through peer-to-peer communication of IP devices

SIP connectivity (also known as SIP trunking) provides a direct media path (trunk interface) between a user in the CS 1000 domain and a user residing in a SIP domain. SIP trunking (the SIP Trunk Gateway) acts as a SIP User Agent and acall-signaling gateway for the telephones (analog [500/2500-type] telephones, digital telephones, and IP Phones). • • As a SIP User Agent, it services one or more end users in making and receiving SIP calls. As a call-signaling gateway, the SIP trunking application does the following: — maps telephony numbers to and from SIP Uniform Resource Identifiers (URIs) — performs client registration — maps ISDNmessages to and from SIP messages

553-3001-213

Standard 3.00

August 2005

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— establishes the speech path between the desktop and SIP endpoints

SIP requests and responses
Table 6 shows the SIP request methods.
Table 6 SIP request methods (Part 1 of 2) Method INVITE Description Indicates a user or service is being invited to participate in a callsession. A re-INVITE message is an INVITE message that is used after a call is answered. Confirms that the client has received a final response to a request. Terminates a call and can be sent by either the caller or the called party. Cancels any pending searches but does not terminate a call that has already been accepted. Queries the capabilities of servers. Provides a mechanism allowing the partysending the message to be notified of the outcome of the referenced request. This can be used to enable many applications, including call transfer. Allows a client to update parameters of a session (such as the set of media streams and their codecs) but has no impact on the state of a dialog. In that sense, it is like a re-INVITE message, but unlike re-INVITE, it can be sent before the initial...