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, yet this functionality often requires digital signal processing tasks that are often costly, resource-intensive and can affect the quality of the voice signals if it introduces too much latency and delay.

The D150 card allows to convert 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 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality. All codecs are fully indemnified; no additional licensing is required for their use.

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, yet this functionality often requires digital signal processing tasks that are often costly, resource-intensive and can affect the quality of the voice signals if it introduces too much latency and delay.

The D150 card allows to convert 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 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality. All codecs are fully indemnified; no additional licensing is required for their use.

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, yet this functionality often requires digital signal processing tasks that are often costly, resource-intensive and can affect the quality of the voice signals if it introduces too much latency and delay.

The D150 card allows to convert 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 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality. All codecs are fully indemnified; no additional licensing is required for their use.

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, yet this functionality often requires digital signal processing tasks that are often costly, resource-intensive and can affect the quality of the voice signals if it introduces too much latency and delay.

The D150 card allows to convert 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 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality. All codecs are fully indemnified; no additional licensing is required for their use.

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, yet this functionality often requires digital signal processing tasks that are often costly, resource-intensive and can affect the quality of the voice signals if it introduces too much latency and delay.

The D150 card allows to convert 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 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality. All codecs are fully indemnified; no additional licensing is required for their use.

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, yet this functionality often requires digital signal processing tasks that are often costly, resource-intensive and can affect the quality of the voice signals if it introduces too much latency and delay.

The D150 card allows to convert 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 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality. All codecs are fully indemnified; no additional licensing is required for their use.

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, yet this functionality often requires digital signal processing tasks that are often costly, resource-intensive and can affect the quality of the voice signals if it introduces too much latency and delay.

The D150 card allows to convert 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 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality. All codecs are fully indemnified; no additional licensing is required for their use.

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, yet this functionality often requires digital signal processing tasks that are often costly, resource-intensive and can affect the quality of the voice signals if it introduces too much latency and delay.

The D150 card allows to convert 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 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality. All codecs are fully indemnified; no additional licensing is required for their use.

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, yet this functionality often requires digital signal processing tasks that are often costly, resource-intensive and can affect the quality of the voice signals if it introduces too much latency and delay.

The D150 card allows to convert 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 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality. All codecs are fully indemnified; no additional licensing is required for their use.

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, yet this functionality often requires digital signal processing tasks that are often costly, resource-intensive and can affect the quality of the voice signals if it introduces too much latency and delay.

The D150 card allows to convert 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 30, 60, 120, 240 or 400 channels of any-to-any voice codec conversion, with unmatched quality. All codecs are fully indemnified; no additional licensing is required for their use.

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.

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.

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.

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.

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.

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.

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

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.

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

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.

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

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.

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

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.

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