Sangoma D100: 30-400 Sessions
- 30 to 480 transcoding sessions
- Intergrates in Asterisk and
- Compact 2U from factor
- PCI and PCIe
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 480 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.
- Call Centers and Remote Agent Pools
- Hosted iPBX / IP
- Distributed PBX
- IP Network Peering
- SIP Trunking
- VoIP Gateways