Chapter 1: Overview Of The Vcx V7122; Introduction - 3Com VCX V7122 SIP User Manual

C 1: O VCX
HAPTER VERVIEW OF THE
V7122

Introduction

The 3Com VCX V7122/SIP (Session Initialization Protocol) VoIP Gateway provides excellent
voice quality and optimized packet voice streaming over IP networks. The VCX V7122/SIP
Gateway enables voice, fax, and data traffic to be sent over the same IP network.
Based on the 3Com award-winning, field-proven TrunkPack design, the VCX V7122/SIP
Gateway uses 3COM DSP voice compression technology.
The VCX V7122/SIP incorporates 1, 2, 4, 8, 16 E1, or T1 spans for connection, directly to
PSTN telephony trunks, and includes one or two 10/100 Base-T Ethernet ports for
connection to the LAN.
The VCX 7122/SIP supports up to 480 simultaneous Voice over IP (VoIP) or Fax over IP
(FoIP) calls, supporting various ISDN PRI protocols such as EuroISDN, North American NI2,
Lucent 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 VCX V7122/SIP 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 VCX V7122/SIP Gateways in branch or
departmental offices, connected to local PBXs and call routing is performed by a centralized
SIP proxy. Thus, the VCX V7122/SIP VoIP Gateway enables you to make low cost long
distance or international telephone/fax calls between distributed company offices, using their
existing PBXs and telephones/fax. These calls are routed over existing IP Internet or Intranet
corporate data network(s). In addition, low bit-rate vocoders and silence suppression can be
used, ensuring that voice traffic takes the minimum of space on the data network.
The VCX V7122/SIP 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 (for example, a call center) is
performed by the VCX V7122/SIP internal routing feature or by a SIP proxy.
Both of the above applications are depicted in Figure 3 on page 15.
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3Com VCX V7122/SIP VoIP Gateway User Manual