Sip Overview - AudioCodes Mediant 600 User Manual

User's Manual
supports up to six analog modules, each module providing four analog RJ-11 ports.
The FXO module can be used to connect analog lines of an enterprise's PBX or the
PSTN to the IP network. The FXS module can be used to connect legacy telephones,
fax machines, and modems to the IP network. Optionally, the FXS module can be
connected to the external trunk lines of a PBX. When deployed with a combination of
FXO and FXS modules, the device can be used as a PBX for Small Office Home
Office (SOHO) users, and businesses not equipped with a PBX.
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Media Processing Module (MPM): The MPM module provides IP media channels for
conferencing and media server functionality. The device can house up to three MPM
modules.
The device has enhanced hardware and software capabilities to ease its installation and to
maintain voice quality. If the measured voice quality falls beneath a pre-configured value,
or the path to the destination is disconnected, the device assures voice connectivity by
'falling' back to the PSTN. In the event of network problems or power failures, calls can be
routed back to the PSTN without requiring routing modifications in the PBX. Further
reliability is provided by dual Ethernet ports and an optional dual AC power supply.
The device supports various ISDN PRI protocols such as EuroISDN, North American NI2,
Lucent™ 4/5ESS, Nortel™ DMS-100 and others. It also supports various ISDN BRI
protocols such as ETSI 5ESS and QSIG over BRI. It also supports different variants of
CAS protocols for E1 and T1 spans, including MFC R2, E&M immediate start, E&M delay
dial / start, loop start and ground start.
The device provides a variety of management and provisioning tools, including an HTTP-
based embedded Web server, Telnet, Element Management System (EMS), and Simple
Network Management Protocol (SNMP). The user-friendly, Web interface provides remote
configuration using a Web browser (such as Microsoft™ Internet Explorer™).
1.3

SIP Overview

Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol used on
the gateway for creating, modifying, and terminating sessions with one or more
participants. These sessions can include Internet telephone calls, media announcements,
and conferences.
SIP invitations are used to create sessions and carry session descriptions that enable
participants to agree on a set of compatible media types. SIP uses elements called Proxy
servers to help route requests to the user's current location, authenticate and authorize
users for services, implement provider call-routing policies and provide features to users.
SIP also provides a registration function that enables users to upload their current locations
for use by Proxy servers. SIP implemented in the gateway, complies with the Internet
Engineering Task Force (IETF) RFC 3261 (refer to http://www.ietf.org).
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1. Overview
Mediant 600 & Mediant 1000