Transcodificação

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, 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.

The TC400B is a bundle of the half-length, low-profile PCI-2.2 compliant TC400P base card and the TC400M voice processing module. The TC400B is designed to handle, in dedicated DSP resources, the complex codec translations for highly compressed audio as would otherwise be processed by Asterisk in software.

Asterisk, in software and with Digium G.729a licensing, is capable of transforming the G.729a codec into other codecs for the purposes of call origination or termination, bridging disparate calls, or VoIP to TDM connectivity. These transformations in software are very expensive, in terms of MIPS, and require a substantial amount of CPU time to accomplish. The TC400B not only relieves the CPU of this duty, freeing it up to handle other tasks or to complete additional call processing; but also provides Asterisk with the capability of bridging G.723.1 compressed audio into other formats, a capability not previously possible.

The TC400B decompresses G.729a (8.0kbit) or G.723.1 (5.3kbit) into u-law or a-law; or, compresses u-law or a-law into G.729a (8.0kbit) or G.723.1 (5.3kbit). The TC400B is rated to handle up to 120 bi-directional G.729a transformations or 92 bi-directional G.723.1 transformations. The TC400B does not require additional licensing fees for the use of these codecs nor does it require the registration process associated with Digium's software-based G.729a codec licensing.

As placas Digium da série TE são de alto desempenho e custo efetivo, com interfaces telefônicas digitais que suportam os ambientes T e E. Os ambientes são seleccionáveis em uma base por cartão ou por porta. Esta característica permite a tradução da sinalização entre os equipamentos T1 e E1, e permite conectar bancos de canais econômicos T1 com circuitos E1.

Às vezes, pode acontecer que o seu sistema de comunicações Ip seja ecoado, isso pode acontecer como resultado dos tempos de espera que um sistema VoIP geralmente tem, ao contrário de um sistema analógico. Como resultado, sua conversa pode sofrer um eco.

Embora muitos dispositivos já incorporem o cancelador de eco, a Digium oferece o hardware de cancelamento de eco de alto desempenho da Digium (HPEC). O hardware de cancelamento de eco também é vantajoso ao lidar com grandes volumes de chamadas ou um grande número de canais que, de outra forma, sobrecarregariam a CPU, o que resultaria em baixa qualidade potencial de áudio. Aqui estão alguns de seus recursos:

1. Octasic baseado em DSP (todos os módulos)
2. 128 ms (1024 taps) de cancelamento de eco em todos os canais
3. G.168 de Qualidade do número certificado de AT & T que está em conformidade com o algoritmo
4. Processador Dinâmico Não-Linear
5. Noise Comfort Generator
6. Pesquisa automática de linhas
7. Cancelar múltiplas reflexões
8. Dupla detecção de diálogos

Os Gateways Digium VoIP oferecem o melhor valor agregado para a conexão de telefonia tradicional (T1/E1/PRI) para IP (SIP).

O software de gateway é baseado no Asterisk é gerenciado por meio de uma interface gráfica do usuário (GUI), que permite fácil navegação e configuração. Eles têm um design integrado de economia de energia com um processador de sinal digital (DSP) altamente eficiente, permitindo excelente manuseio de todas as operações relacionadas à mídia.

The TCE400B is a bundle of the half-length, low-profile PCI-Express x1 TCE400P base card and the TC400M voice processing module. The TCE400B is designed to handle, in dedicated DSP resources, the complex codec translations for highly compressed audio as would otherwise be processed by Asterisk in software.

Asterisk, in software and with Digium G.729a licensing, is capable of transforming the G.729a codec into other codecs for the purposes of call origination or termination, bridging disparate calls, or VoIP to TDM connectivity. These transformations in software are very expensive, in terms of MIPS, and require a substantial amount of CPU time to accomplish. The TCE400B not only relieves the CPU of this duty, freeing it up to handle other tasks or complete additional call processing, but also provides Asterisk with the capability of bridging G.723.1 compressed audio into other formats, a capability not otherwise possible.

The TCE400B decompresses G.729a (8.0kbit) or G.723.1 (5.3kbit/6.0kbit) into G.711 u-law or a-law; or compresses G.711 u-law or a-law into G.729a (8.0kbit) or G.723.1 (5.3kbit). The TCE400B is rated to handle up to 120 bi-directional G.729a transformations or 92 bi-directional G.723.1 transformations. The TCE400B does not require additional licensing fees for the use of these codecs nor does it require the registration process association with Digium's software-based G.729a codec licensing.

As placas Digium da série TE são de alto desempenho e custo efetivo, com interfaces telefônicas digitais que suportam os ambientes T e E. Os ambientes são seleccionáveis em uma base por cartão ou por porta. Esta característica permite a tradução da sinalização entre os equipamentos T1 e E1, e permite conectar bancos de canais econômicos T1 com circuitos E1.

Às vezes, pode acontecer que o seu sistema de comunicações Ip seja ecoado, isso pode acontecer como resultado dos tempos de espera que um sistema VoIP geralmente tem, ao contrário de um sistema analógico. Como resultado, sua conversa pode sofrer um eco.

Embora muitos dispositivos já incorporem o cancelador de eco, a Digium oferece o hardware de cancelamento de eco de alto desempenho da Digium (HPEC). O hardware de cancelamento de eco também é vantajoso ao lidar com grandes volumes de chamadas ou um grande número de canais que, de outra forma, sobrecarregariam a CPU, o que resultaria em baixa qualidade potencial de áudio. Aqui estão alguns de seus recursos:

1. Octasic baseado em DSP (todos os módulos)
2. 128 ms (1024 taps) de cancelamento de eco em todos os canais
3. G.168 de Qualidade do número certificado de AT & T que está em conformidade com o algoritmo
4. Processador Dinâmico Não-Linear
5. Noise Comfort Generator
6. Pesquisa automática de linhas
7. Cancelar múltiplas reflexões
8. Dupla detecção de diálogos

Os Gateways Digium VoIP oferecem o melhor valor agregado para a conexão de telefonia tradicional (T1/E1/PRI) para IP (SIP).

O software de gateway é baseado no Asterisk é gerenciado por meio de uma interface gráfica do usuário (GUI), que permite fácil navegação e configuração. Eles têm um design integrado de economia de energia com um processador de sinal digital (DSP) altamente eficiente, permitindo excelente manuseio de todas as operações relacionadas à mídia.

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

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