Building block for deploying high-density VoIP systems

The TP-1610 cPCI VoIP communication blade, based on dual AudioCodes’ TPM-1100 PMC Modules, is an ideal building block for deploying high-density, high-availability Voice over IP (VoIP) systems. The TP-1610 is suitable for VoIP gateways, IP-enabled call centers, large telephony companies and next generation DLCs. Offering integrated voice gateway functionality capable of delivering up to 480 simultaneous calls, the TP-1610 supports all necessary functions for voice and fax streaming over IP networks.

Deliver Feature-Rich Solutions
The TP-1610 supports a broad selection of voice-processing-related algorithms, including G.711, G.723.1 and G.729AB Vocoders, G.168-compliant echo cancellation, T.38 real-time fax over IP, and a wide selection of in-band and out-band tone detection and generation. In addition, signaling protocols supported include ISDN PRI, SIGTRAN (M2UA, M3UA, IUA) and CAS.

Comply with Industry Standards
The TP-1610 blade complies with industry standard network control protocols including MGCP, MEGACO (H.248), SIP, H.323 as well as AudioCodes’ proprietary TPNCP. These allow for the implementation of a distributed gateway architecture that separates call processing functions from media streaming functions, resulting in better redundancy, scalability and higher system availability.

Protect Customer Investment
The TP-1610 is based on the VoIPerfect™ architecture, AudioCodes’ underlying, best-of-breed, core media gateway technology for all of its products. The TP-1610 supports AudioCodes’ API, which enables software download, provisioning and control. It was designed to maintain essential API backward compatibility in order to protect customers’ investment in the development of products based on former generations.

Enable Fast and Easy Integration
Enabling accelerated design cycles with high 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 migration to VoIP networks.

    • High channel density
    • Concurrent toll-quality voice and fax support
    • Carrier-grade applications
    • Reduced system cost and increased reliability
    • Fast time-to-market
    • Flexible and easy migration to VoIP networks
    • Extensive VoIP experience
    • Up to 480 voice/fax independent multiple LBR channels
    • VoIP packet streaming (RTP/RTCP) per RFC 3550/3551
    • Standard control: MGCP (RFC 2705), MEGACO (H.248), SIP/H.323
    • Real-time fax over IP/T.38
    • On-board announcement support towards PSTN/TDM and IP
    • Tone detection and generation (MF, DTMF, RFC 2833)
    • PSTN Signaling: CAS, ISDN PRI, V5.2 (AN), and SS7
    • SIGTRAN IUA, M2UA, M3UA over SCTP
    • cPSB PICMG 2.16 compliant ethernet on the backplane
  • Capacity 60, 120, 240 or 480 independent digital voice, fax and data ports
    Voice Compression G.711, G.723.1, G.729A/B, G.726/G.727, NetCoder• MS-GSM, GSM-FR, iLBC Additional coders supported – contact AudioCodes for further information
    Echo Cancellation G.168 compliant 32, 64, 128 msec echo tail
    Gain Control Programmable
    Fax Relay and Termination Real-time fax over IP/T.38 compliant, automatic fallback to G.711 and VBD for up to super G-3 fax machines
    Support for Fax Termination (Available with AudioCodes S/W based stack)
    ASR-third-party Recognition Engines Host-based Architecture – Media Stream over PCI
    Distributed Architecture – Media Stream over VoIP RTP
    In-band/Out-of-band Signaling Packet side or PSTN side, DTMF and tone detection and generation
    IVR Support On-board announcement storage – 10 Mb
    Recorded prompts – 20 minutes of G.711, 200 minutes of G.723
    VoIP Standards Compliance RTP/RTCP per RFC 3550/3551
    DTMF over RTP per RFC 2833
    Control Protocols Media Gateway on a blade mode:
    Controlled by either MGCP or MEGACO
    PCI used for power only
    SIP, H.323
    TPNCP – AudioCodes’ proprietary VoIP API Library
    Management Interfaces
    • SNMP V2c: Standard MIB-2: system, interfaces, if-MIB, entity-MIB, RTP-MIB, DS1-MIB, snmpV2-MIB and AudioCodes’ proprietary MIB
    • On-board embedded secure Web Server
    Operating System
    • Windows™ 2000, XP, 2003
    • Linux™ RH8, RH9, Debian, Enterprise
    • Solaris™ 8,9 on Intel™/Sparc™ 32/64

    Signaling

    PSTN CAS T1 robbed bit, MFC/R2 numerous country variants
    CCS ISDN PRI: numerous country variants including ETSI EURO ISDN, ANSI NI2, DMS, 5ESS, Japan INS1500 V5.2 AN (Contact AudioCodes)
    SS7 MTP2 and MTP3 link termination ISUP and SCCP/TCAP termination (Available with AudioCodes S/W based stack)
    SIGTRAN M2UA, M3UA, IUA and DUA over SCTP per RFC 2960

    Hardware Specifications

    Ethernet Dual redundant 100 BASE-T ports
    Hot Swap Full hot swap supported per PICMG 2.1
    Physical Interfaces Form factor – 6U PICMG 2.0 single cPCI slot
    TDM Interface – H.110 CT Bus
    Telephony – two 50-pin Telco connectors on rear panel
    Ethernet – cPSB PICMG 2.16 on the backplane, Dual RJ-45 on rear panel
    Power 40.7W, 3A at 5V, 7.8A at 3.3V