Version 6.6 19 Mediant 2000
User's Manual 1. Overview
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 VoIPerfectâ„¢ voice
compression technology.
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, Lucentâ„¢ 4/5ESS, Nortel
â„¢
DMS-100 and others.
In 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 calls between distributed company offices, using their existing
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:
• 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' on page 697.
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