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Tarjeta Sangoma D100-120E transcoding 120chan PCIe

  • D100: 30-400 Sessions

    Voice Transcoding Boards

    The industry’s first affordable PCI or PCI Express voice transcoding card — designed for optimum voice quality for low to high density system!

    • 30 to 400 transcoding sessions
    • Field Upgradeable
    • No Licensing Fees²
    • Works on Asterisk® and FreeSWITCH®
    • Reduce Host CPU Load
    • Simple API for Application Integration
    • Integrated or in Dedicated Server
    • Available in PCI or PCIe
    • Small Foot Print

    Overview:

    IP telephony applications commonly require the use of multiple voice codecs, used to digitally compress voice signals and save on bandwidth. Voice signals from the Public Switched Telephone Network (PSTN) come in the form of the G.711 codec, but the VoIP terminal equipment and networks can support a variety of different voice codecs, such as G.729, G.726, AMR, G.723.1, G.722, iLBC, etc. The VoIP infrastructure needs the capability to mediate between endpoints supporting different codecs, but this functionality requires digital signal processing tasks that are often costly and resource-intensive, and can affect the quality of the voice signals, if it introduces too much latency and delay.

    The D100 card, available in PCI and PCI Express form factors, converts simultaneous channels of transcoding from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. The card allows up to 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality¹. All codecs are fully indemnified; no additional licensing is required for their use².

    The D100 works with both Asterisk and FreeSWITCH. With compatible drivers, these applications can use the D100 cards as seamless voice transcoding resources.

    transcoding_diagram

    Sample Applications:

    • Call Centers and Remote Agent Pools
    • Hosted iPBX / IP Centrex
    • Distributed PBX
    • IP Network Peering
    • SIP Trunking
    • VoIP Gateways

    ¹ Total port counts performance varies from codec to codec use and on the traffic mix at any given time.

    ² Except for AMR and AMR-WB

    • Codecs SoportadosG.711, G.722, G.722.1, G.723.1, G.726, G.729AB, GSM-FR, GSM-EFR, AMR, AMR-WB (G.722.2), iLBC, L8 (Linear 8K), L16 (Linear 16K)
    • Caracteristicas AdicionalesSoporte para Asterisk y FreeSWITCH
    • Numero de llamadas simultaneasHasta 120
    • ConectorPCI express
    • Tamaño94 mm x 63.5 mm. Incluye bracket de perfil bajo para insertar en chasis 2U

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