Cloud-based speech recognition and conversational engines provided by Google.
Cloud-based speech recognition and synthesis engines provided by Microsoft.
Cloud-based speech recognition and synthesis engines provided by Amazon.
Cloud-based speech recognition and synthesis engines provided by Nuance.
Cloud-based speech recognition and synthesis engines provided by IBM.
Cloud-based speech recognition and synthesis engines provided by Yandex.
Speech recognition and synthesis engines provided by NVIDIA.
Speech recognition and synthesis engines provided by SoundHound.
Speech recognition and synthesis engines provided by Verbio.
Genesys Voice Platform (GVP) is a software-only, standards-based voice portal that provides cost-effective customer interactions for businesses using voice, video, the web, and the cloud.
MoreAvaya Aura Experience Portal is the latest generation of Avaya Voice Portal. Experience Portal provides organizations with a single point of orchestration of all automated voice and multimedia applications and services.
MoreCisco® Unified Contact Center Express (CCX) helps businesses and organizations deliver a connected digital experience, enabling contextual, continuous, and capability-rich journeys.
MoreFreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media.
MoreAsterisk is an open source framework for building communications applications. Asterisk powers IP PBX systems, VoIP gateways, conference servers and other custom solutions.
MoreUniMRCP is an open source cross-platform implementation of the MRCP client and server in the C/C++ language distributed under the terms of the Apache License 2.0.
MoreUniMRCP integration allows Asterisk to facilitate the use of the industry standard media resource servers for implementation of speech applications.
MoreFreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many applications can be developed using a wide range of free tools.
FreeSWITCH can perform full video transcoding and MCU functionality using its conferencing module. FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols. FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 12, 16, 24, 32 or 48 kilohertz in mono or stereo and can bridge channels of different rates.
FreeSWITCH can be used with UniMRCP server in order to utilize a big variety of speech recognition and synthesis engines either installed on-premise or available as service.
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