3Com TP-1610, Руководство пользователя

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V6100 and V7122 User Guide
The V7122 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 V7122
internal routing feature or by a SIP Proxy.
The V7122 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 V7122 as a 1-module
platform and must be repeated for the second module as well.
TP-1610 Overview
The TP-1610 cPCI VoIP media gateway board, based on dual TPM-1100 PMC Modules, is a
complete SIP-compliant ‘two media gateways on a board’, delivering cost-effective solution
in a convenient cPCI form-factor.
The TP-1610 is an ideal solution for SIP trunking gateways and integrated media gateways
for IP-PBXs and all-in-one communication servers. The board is designed for enterprise or
carrier applications. The TP-1610 provides up to 480 simultaneous ports for voice, fax or
data for VoIP media gateway applications providing excellent voice quality and optimized
packet voice streaming over IP networks. Employing SIP as a control protocol, the TP-1610
enables vendors and System Integrators (SIs) short time-to-market and reliable cost-
effective deployment of next-generation networks.
One or two packet processors (depending on the board's capacity) handle packet-streaming
functions through two, redundant integral 10/100 Base-TX interfaces. Each processor
implements the industry-standard RTP/RTCP packet-streaming protocol, advanced adaptive
jitter buffer management, and T.38 fax relay over IP.
The TP-1610 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 TP-1610 enables 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 TP-1610 internal routing feature or by a SIP Proxy.
Enabling accelerated design cycles with higher density and reduced costs, the TP-1610 is an
ideal building block for scalable, reliable VoIP solutions. With the TP-1610’s comprehensive
feature set, customers can quickly design a wide range of solutions for PSTN and VoIP
networks.