Sip Overview - AudioCodes Mediant 1000 User Manual

The Mediant 1000 has enhanced hardware and software capabilities to ease its installation
and to help maintain voice quality. If the measured voice quality falls beneath a pre-
configured value, or the path to the destination is disconnected, the Mediant 1000 can
assure 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 optional dual AC
power supply.
The Mediant 1000 supports various ISDN PRI protocols such as EuroISDN, North
American NI2, Lucent™ 4/5ESS, Nortel™ DMS100 and others. In addition, it 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 Mediant 1000 provides a user-friendly embedded HTTP-based Web server for remote
configuration and management using a standard Web browser (such as Microsoft™
Internet Explorer™ or Netscape™ Navigator™), from anywhere in the world with IP
connectivity to the device.
1.1

SIP Overview

Session Initialization 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.)
SIP User's Manual 20
Mediant 1000
Document #: LTRT-83302