Overview, 1 overview – Nortel Networks Mediant TP-1610 SIP User Manual

Page 13

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Mediant 2000 SIP User’s Manual

1. Overview

Version 4.4

13

July 2005

1 Overview

The Mediant 2000 SIP

Voice over IP (

VoIP) gateway enables voice, fax, and data traffic to be sent

over the same IP network. The Mediant 2000 provides excellent voice quality and optimized
packet voice streaming over IP networks.

The Mediant 2000 uses the award-winning, field-proven Digital Signal Processing (DSP) voice
compression technology used in other TrunkPack

TM

series products.

The Mediant 2000 incorporates 1, 2, 4, 8 or 16 E1 or T1 spans for connection, directly to Public
Switched Telephone Network (PSTN) / Private Branch Exchange (PBX) telephony trunks, and
includes one or two 10/100 Base-TX Ethernet ports for connection to the network.

The Mediant 2000 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

DMS100 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 Mediant 2000 gateway, 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 Mediant 2000 gateways in branch or
departmental offices, connected to local PBXs. Call routing is performed by the gateways
themselves or by SIP Proxy(s).

The Mediant 2000 gateway enables Users to make low cost long distance or international
telephone/fax calls between distributed company offices, using their existing telephones/fax.
These calls are routed over the existing network ensuring that voice traffic uses minimum
bandwidth.

The Mediant 2000 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 media gateway. This provides the ability to transmit voice and
telephony signals between a packet network and a TDM network. Routing of the calls from the
PSTN to a SIP service node (e.g., Call Center) is performed by the Mediant 2000 internal routing
feature or by a SIP Proxy.

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