![]() ![]() The 180 Ringing response indicates that the user is being alerted. The Cisco SIP IP phone sends a SIP 180 Ringing response to Gateway 1. The 100 Trying response indicates that the INVITE request has been received by the Cisco SIP IP phone.ġ80 Ringing-Cisco SIP IP phone to Gateway 1 The Cisco SIP IP phone sends a SIP 100 Trying response to Gateway 1. Gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.ġ00 Trying-Cisco SIP IP phone to Gateway 1 The port on which the Gateway is prepared to receive the RTP data is specified.The media capability User A is ready to receive is specified.The transaction number within a single call leg is identified in the CSeq field.A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.PBX A is identified as the call session initiator in the From field.The IP address of the Cisco SIP IP phone is inserted in the Request-URI field.Gateway 1 sends a SIP INVITE request to the address it receives as the dial peer, which, in this scenario, is the Cisco SIP IP phone. The dial peer includes the IP address and the port number of the SIP-enabled entity to contact. Gateway 1 maps the SIP URL phone number to a dial peer. The Call Setup includes the standard transactions that take place as User A attempts to call User B. Gateway 1 is connected to the Cisco SIP IP phone over an IP network.Ĭall Setup is initiated between PBX A and Gateway 1. User B is located at a Cisco SIP IP phone. PBX A is connected to Gateway 1 (SIP Gateway) via a T1/E1. In this scenario, the two end users are User A and User B. Call Flow between PBX to Cisco SIP IP Phone-Successful Setup and Disconnectīelow diagram illustrates a successful gateway-to-Cisco SIP IP phone call setup and disconnect. Basic knowledge about the SIP Protocol and the call flow Messages.ġ.Also this document covers the SIP Troubleshooting commands. Idea of creating this document is to help the beginners to understand the Various SIP Call flows and messages. This document explains the basic SIP Call flow between the PBX, Gateways and SIP Phones in detail. ![]()
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