Session Initiation Protocol (Sip) Overview - Cisco ATA-186 Administrator's Manual

Session Initiation Protocol (SIP) Overview

Figure 1-2
Telephone/fax
Figure 1-3
Telephone/fax
Session Initiation Protocol (SIP) Overview
Session Initiation Protocol (SIP) is the Internet Engineering Task Force (IETF) standard for real-time
calls and conferencing over Internet Protocol (IP). SIP is an ASCII-based, application-layer control
protocol (defined in RFC3261) that can be used to establish, maintain, and terminate multimedia
sessions or calls between two or more endpoints.
Like other Voice over IP (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 for the Cisco ATA is compliant with RFC2543.
Note
This section contains the following topics:
•
•
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide (SIP)
1-2
Cisco ATA 186 as Endpoint in SIP Network
V
Ethernet
Cisco ATA 186
Cisco ATA 188 as Endpoint in SIP Network
V Ethernet
Cisco ATA 188
SIP Capabilities, page 1-3
Components of SIP, page 1-3
Chapter 1
Broadband
Broadband CPE
(DSL, cable,
fixed wireless)
Broadband
Broadband CPE
(DSL, cable,
fixed wireless)
Cisco Analog Telephone Adaptor Overview
Voice
Layer 3
gateway
IP infrastructure
V
SIP proxy
Voice
Layer 3
gateway
IP infrastructure
PSTN
V
SIP proxy
OL-3410-01
PSTN