Enhanced voice service applications

The IPM-6310 (IPmedia) cPCI VoIP Media Processing blade is a high-density, high-performance VoIP media processing blade designed for enhanced voice service applications for both existing and next generation networks. Introducing a comprehensive set of advanced media processing and signaling capabilities on a single cPCI blade, the IPM-6310 is an excellent building block for media services and VoIP application developers targeting better flexibility and short time-to-market.

Deliver Feature-rich Solutions
A wide selection of firmware-based media processing capabilities is available with the IPM-6310 including: message record/playback, conferencing, onboard announcement storage, IVR streaming and control, voice coding and transcoding, echo cancellation, fax processing and call progress tone detection. Each channel resource on the IPM-6310 is universal and can perform media processing functions independently and simultaneously while utilizing flexibility of endpoints.

Comply with Industry Standards
The IPM-6310 blade complies with industry standard network control protocols including SIP, MEGACO (H.248), MGCP and AudioCodes’ proprietary API – VoPLIB. This allows the implementation of a distributed media server architecture that separates call processing functions from media processing functions. The blade enables scalability, better redundancy and higher system availability.

Protect Customer Investment
The IPM-6310 is based on VoIPerfect™ architecture, AudioCodes’ field-proven, best-of-breed, core media gateway technology for all of its products. The blade supports AudioCodes’ API (VoPLIB) 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 prior generations.

Enable Fast and Easy Integration
The IPM-6310 is a foundation for scalable, reliable VoIP enabled media processing solutions and enables accelerated design cycles with high-density and reduced costs. The comprehensive feature set of the IPM-6310 allows customers to quickly design a wide range of solutions combining VoIP and PSTN networks.

    • Provides high-density, high-performance blade for carrier grade VoIP applications
    • Enhanced IP-enabled media service features
    • Embodies rich and comprehensive API (VoPLIB)
    • Implements field-proven and cost-effective technology
    • Enables scalable distributed architecture
    • Reduces development cycle
    • Up to 2016 universal media processing ports
    • Comprehensive IVR control
    • Real-time, multi-party conferencing with mixed IP/PSTN/H.110 endpoints
    • High-density voice record/playback over the network
    • Real-time fax over IP/T.38
    • Voice transcoding between wireline and wireless networks
    • VoIP packet streaming (RTP/RTCP) per RFC 3550/3551
    • SIP, MEGACO, MGCP and AudioCodes’ proprietary API (VoPLIB)
    • CPSB PICMG 2.16 compliant Ethernet on the backplane
    • Optional STM-1/OC-3 or 3 x T3 PSTN interfaces
  • Voice Messaging, Recording Record/play using standard NFS streaming – 2016 channels
    On-board announcement storage – 32 MB: 60 min of G.711
    RTP forking replications for lawful intercept (CALEA)
    N-way Conferencing Supports up to 2016 ports of Mixed IP, PSTN and TDM (H.110) participants
    Maximum full-duplex parties per conference bridge: 64 endpoints
    Up to 3 simultaneous active participants or linear summation mode (all active)
    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 SG-3
    Concurrent fax sessions on all the channels
    Support for Fax Termination (Available with AudioCodes S/W based stack)
    ASR Distributed Architecture – Media Stream over VoIP RTP
    Voice Compression Supports up to 2016 ports of:
    G.711, G.723.1, G.729A/B, G.726, G.727, GSM-FR, GSM-EFR, EVRC, NB-AMR, iLBC
    Wide Band coders including G.722 and AMR
    Additional coders supported
    Voice Processing Voice Activity Detection (VAD) and CNG
    Transcoding of LBR to LBR coder supporting 1024 channels
    Transcoding of G.711 RTP to any LBR coder supporting 2048 channels
    Echo Cancellation G.168 with tail of 32 msec, 64 msec and 128 msec
    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 Protocol AudioCodes’ proprietary API - VoPLIB, SIP, MEGACO (H.248), MGCP (RFC 3435, RFC 3660)
    Management
    • 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
    SIGTRAN M2UA, M3UA, IUA and DUA over SCTP per RFC 2960

    Hardware Specifications

    Form Factor 6U PICMG 2.16 compliant, single cPCI slot, full Hot Swap
    Interfaces Dual GBEth, TDM: H.110, PSTN: OC-3/STM-1 APS protected or 3 x T3
    Power 40-85 W