Voice and Video convergent IVR platform
Our IVVR (or Video IVR) Server is one of the few voice and video convergent VoiceXML platform in the market. It works with voice calls as well as video calls to deploy today amazing next generation communication services. Its add-on implements many features to play, record, convert or transcode video calls from different channels to many others. Make voice and video calls from any browser and over any IP network. This Video IVR solution works from everywhere, without any plugin and is interoperable with SIP or RTMP voice and video channels.
- VoiceXML 2.1 with many extended tags added
- VoiceXML extended tags (like ,…)
- Call Control functions/variables for Asterisk
- High performance C built
- Application for Asterisk PBX (app_vxml.so)
- Binaries available for Linux OS 32bit and 64bit
- VoiceXML accounts management
- Support plugging objects with VoiceXML tag
- VoIP / SS7 / ISDN / RTMP Networks
- Fax over VoiceXML (send and receive)
- Text-to-Speech (TTS) connectors
- Automatic-Speech-Recognition (ASR) connectors
- MRCP v1 and v2 (multi-grammar allowed)
- Voice-Silent Detection (VSD)
- Support RTSP-URI with
Voice & Video ConvergentOur approach is a voice and video convergent platform for Asterisk. It works with voice calls as well as video calls to deploy today amazing next generation communication services.
Text-to-VideoWe supports a Text-to-Video (TTV) extended feature for video services enabling to generate a real-time video content from a text in order to print a message or dynamic menu on your video phone screen.
Text-to-SpeechWe support most Text-to-Speech (TTS) engine thru an advanced, universal HTTP and MRCP open source interface that helps developers manage multi-vendor TTS engines in the same or different server(s).
Wifi or 3G/4G NetworksOur Video IVR solution is ready for many networks like public Wifi or Cellular. It’s not limited in bandwidth conditions, NAT or ports usage. You can use it from everywhere on over with any carrier network.
Multi-ProtocolWe are running many telephony audio and video protocols like SIP, RTMP, H323,…. We do not require any particular skills in terms of architecture or proprietary programming.
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