Building Block for deploying voice over packet (VoP) systems

The TrunkPack 6310 VoIP communication platform, with a STM-1/OC-3 PSTN interface, is an ideal building block for deploying high-density, high-availability Voice over Packet (VoP) systems. The TP-6310 is suitable for high-density VoIP gateways, packet-to-packet mediation, media servers and cable telephony gateways. Offering integrated voice and signaling gateway functionality, the TP-6310 supports all necessary functions for voice and fax streaming over IP networks.

Deliver Feature-Rich Solutions
The TP-6310 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, a wide selection of in-band and out-of-band tone detection and generation. In addition, the TP-6310 also supports signaling protocols including ISDN PRI, SIGTRAN (M2UA, M3UA, IUA) and CAS. All media processing, signaling and control protocols are applied independently and simultaneously on all of the 2016 LBR channels.

Comply with Industry Standards
The TP-6310 blade complies with industry-standard network control protocols including MGCP, MEGACO (H.248) as well as AudioCodes’ proprietary API (TPNCP). This allows for the implementation of a distributed gateway architecture that separates call-processing functions from media streaming functions. This results in better redundancy, scalability and higher system availability.

Protect Customer Investment
The TP-6310 is based on the VoIPerfect™ architecture, AudioCodes’ underlying, core media gateway technology for all of its products. The TP-6310 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-6310 is an ideal building block for reliable VoP solutions. With the TP-6310’s comprehensive feature set, customers can quickly design a wide range of solutions for migration to VoP networks.

    • Very high channel density – up to 2016 LBR VoIP channels
    • Packet-to-packet mediation
    • Concurrent toll-quality voice and fax support on all channels
    • STM-1/OC-3 Automatic Protection Switching on PSTN interface
    • Designed for carrier-grade applications
    • Built on 4 previous generations of proven and widely deployed VoP technology
    • Flexible and easy migration to VoP networks
    • 2016 voice/fax independent multiple LBR channels
    • Integrated Automatic Protection Switching (APS) OC-3/STM-1 for PSTN interface
    • Standard control: MGCP, MEGACO
    • Complete media gateway on a blade
    • G.168 compliant echo cancellation
    • Real-time fax over IP/T.38
    • PSTN Signaling: CAS, ISDN PRI and SSlayer 3 termination
    • Tone detection and generation (MF, DTMF, RFC 2833)
    • SIGTRAN IUA, M2UA, M3UA over SCTP
    • Dual redundant 10/100/1000 Base-T interfaces, PICMG 2.16 compliant
  • Capacity 2016 independent digital voice, fax and data ports
    Up to 1000 additional mediation channels without transcoding
    Voice Compression G.711, G.723.1, G.729A/B, G.726/G.727, AMR, MS-GSM, GSM-FR, iLBC
    Wide Band coders including G.722 and AMR
    Additional coders supported – contact AudioCodes for further information
    Echo Cancellation G.168 compliant 32, 64, 128 msec echo tail
    Fax Relay and Termination Real-time fax over IP/T.38 compliant, automatic fallback to G.711 and VBD for up to SG-3
    Concurrent fax sessions on all the channels
    Support for Fax Termination (Available with AudioCodes S/W based stack)
    ASR-third-party Recognition Engines Distributed Architecture - Media Stream over VoIP RTP
    In-band/Out-of-band Signaling Packet side or PSTN side, DTMF and tone detection and generation, CAS Relay, RFC 2833
    VoIP Standards Compliance RTP/RTCP per RFC 3550/3551
    DTMF over RTP per RFC 2833
    Control Protocols AudioCodes’ proprietary API - VoPLIB, SIP, MEGACO (H.248), MGCP (RFC 3435, RFC 3660)
    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
    • SysLog
    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
    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 10/100/1000 BASE-TX 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 – Dual optical Replaceable LC Connector, APS protected
    Ethernet – cPSB PICMG 2.16 on the backplane, Dual Optical Replaceable
    LC connectors on rear panel
    Power 75W