Build next-generation applications for today and tomorrow

The IPM-1610 is a complete VoIP media processing solution providing IP and PSTN interfaces as needed to build next-generation applications for both today’s and tomorrow’s networks. Moreover, by introducing a packet interface for voice streaming the blade AudioCodes enables a smooth evolution from legacy PCI bus architecture to distributed, scalable, packet-based architecture.

Deliver Feature-Rich Solutions
A broad selection of firmware-based media processing capabilities is available with the IPM-1610 including: message record/playback, conferencing, on-board announcements storage, voice coding, echo cancellation, fax processing and call progress tone detection. Each channel resource on the IPM-1610 is universal and can perform media processing functions while utilizing full flexibility in endpoints.

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

Protect Customer Investment
The IPM-1610 is based on the VoIPerfect™ architecture, AudioCodes’ underlying, best-of-breed, core media gateway technology for all of its products. The IPM-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 IPM-1610 is an ideal building block for scalable, reliable VoIP enabled media processing solutions. With the IPM-1610’s comprehensive feature set, customers can quickly design a wide range of solutions combining PSTN and VoIP networks.

    • IP-enabled, cost-effective technology
    • Field-proven PSTN interface blade
    • High-density, high-performance blade
    • Independent call-by-call basis LBR ports
    • Carrier grade applications
    • Concurrent toll quality voice and fax support
    • Enables scalable distributed architecture
    • Shorter development cycle
    • Up to 480 IVR streaming ports
    • Up to 240 universal media processing ports
    • Voice record/playback
    • Interchangeable RTP or PSTN or H.110 endpoints
    • Real-time, multi-party conferencing
    • Comprehensive IVR control
    • VoIP packet streaming (RTP/RTCP) per RFC 3550/3551
    • MGCP, MEGACO, SIP and AudioCodes’ proprietary TPNCP
    • cPSB PICMG 2.16 compliant Ethernet on the backplane
    • Automatic Speech Recognition (ASR)
    • Text to Speech (TTS)
  • Configuration “Streaming”– 480 Voice/Fax Messaging ports
    “Universal” – 120, 240 universal ports
    Voice Messaging, Recording Host-based record/play. WAV format (G.711. G.726, MS-GSM
    Fast slow playback with pitch correction
    Timeslot summation – Record RX+TX of the call
    On-board announcement storage – 10 Mb: 20 minutes of G.711, 200 minutes of G.723
    RTP forking replication for lawful intercept (CALEA)
    Recording/playback using standard HTTP or NFS streaming
    N-way Conferencing Supports up to 240 ports of Mixed IP, PSTN and TDM (H.110) participants
    Maximum simultaneous 3-way conferences per blade: 80
    Maximum full-duplex parties per conference bridge: 64 endpoints
    Supports various conference control modes
    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
    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
    Voice Processing Voice Activity Detection (VAD), Comfort Noise Generation (CNG)
    Trans-coding of LBR/HBR to any LBR/HBR stream
    Echo Cancellation G.168 with tail of 30 msec, 64 msec and 128 msec 3
    Gain Control Automatic (AGC) or programmable
    In-band/Out-band Signaling Packet side or PSTN side, DTMF and tone detection and generation, RFC 2833
    Control AudioCodes’ proprietary TPNCP, MGCP (RFC 3435), MEGACO (H.248), SIP
    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™ 2 RH8, RH9, Debian, Enterprise
    • Solaris™ 2 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, SS7 monitoring ISUP and SCCP/TCAP termination (Available with AudioCodes S/W)
    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