Sangoma D500: 400-2,000 Sessions
- 400-2,000 Transcoding Sessions
- No Licensing Fees
- Integrates in Asterisk and
- Works on Linux and Windows
- Simple C API
- PCI Express
Most IP telephony applications require the use of multiple types of voice codecs, which are used to digitally compress voice signals, to save on bandwidth requirements. While voice signals from the Public Switched Telephone Network (PSTN) always come in the form of the G.711 codec, the VoIP terminal equipment and networks support a variety of different voice codecs including such as G.729, G.726, AMR, G.722, iLBC, etc. VoIP infrastructure most often needs to include the capability to mediate between endpoints supporting different codecs, but this functionality often requires digital signal processing tasks that are costly, resource intensive and can affect the quality of the voice signals if it introduces too much latency and delay.
The D500 card converts numerous 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 running up to 2000 sessions of any-to-any voice codec conversion, with unmatched quality1. All codecs are fully indemnified; no additional licensing is required for their use2.
The D500 works with both Asterisk and FreeSWITCH. With compatible drivers offered by Sangoma, these applications can use the D500 cards as seamless voice transcoding resources. Alternatively, developers and integrators can use the Transcoding API in C for their own application development.
- Hosted VoIP
- PBX with HD Voice
- IP Network Peering
- Hosted PBX
- Call Centers