Overview - AudioCodes Mediant 2000 User Manual

SIP User's Manual
1

Overview

This manual provides you with the information for installing, configuring, and operating the
Mediant 2000 SIP gateway (referred to throughout this manual as device).
The device is a SIP-based Voice-over-IP (VoIP) media gateway. the device enables voice,
fax, and data traffic to be sent over the same IP network.
The device provides excellent voice quality and optimized packet voice streaming over IP
networks. The device uses the award-winning, field-proven VoIPerfect™ voice compression
technology, typically implemented in AudioCodes products.
The device incorporates 1, 2, 4, 8 or 16 E1, T1, or J1 spans for direct connection to the
Public Switched Telephone Network (PSTN) / Private Branch Exchange (PBX) through
digital telephony trunks. The device also provides SIP trunking capabilities for Enterprises
operating with multiple Internet Telephony Service Providers (ITSP) for VoIP services. The
device includes two 10/100Base-TX Ethernet ports, providing redundancy connection to the
network.
The device supports up to 480 simultaneous VoIP or Fax over IP (FoIP) calls, supporting
various Integrated Services Digital Network (ISDN) Primary Rate Interface (PRI) protocols
such as EuroISDN, North American NI2, Lucent™ 4/5ESS, Nortel
addition, it supports different variants of Channel Associated Signaling (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, best suited for large and medium-sized VoIP applications is a compact device,
comprising a 19-inch, 1U chassis with optional dual AC or single DC power supplies. The
deployment architecture can include several devices in branch or departmental offices,
connected to local PBXs. Call routing is performed by the devices using internal routing or
SIP Proxy(s).
The device enables users to make cost-effective, long distance or international
telephone/fax
telephones/fax. These calls can be routed over the existing network using state-of-the-art
compression techniques, ensuring that voice traffic uses minimum bandwidth.
The device can also route calls over the network using SIP signaling protocol, enabling the
deployment of Voice over Packet solutions in environments where access is enabled to
PSTN subscribers by using a trunking device. This provides the ability to transmit voice and
telephony signals between a packet network and a TDM network.
Notes:
Version 5.8
calls between distributed
•
The device is offered as a 1-module (up to 240 channels or 8 trunk
spans) or 2-module (for 480 channels or 16 trunk spans only) platform.
The latter configuration supports two TrunkPack modules, each having its
own IP address. Configuration instructions in this document relate to the
device as a 1-module platform and must be repeated for the second
module as well.
•
For channel capacity, refer to the device's specifications in ''Selected
Technical Specifications''
company offices,
on page 491 .
19
1. Overview
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DMS100 and others. In
using their existing
October 2009