Call, Video Call and Chat
WebRTC is direct communication through the browser.
If your customer has access to a browser – Chrome, Firefox or Opera – he’ll be able to communicate with the service center. By calling, video-calling or chatting – but above all encrypted and therefor absolutely safe.
A WebRTC can be directly embedded on a website and offers flexible choice between the communication channels voice, video and chat. It is therefore an easy and attractive plus that gives the user a very comfortable means of communicating with your enterprise.
It is suitable not only for internal corporate communication, but also for direct contact between the customer and a service center.
General service specification
Audio- and video-calls and live-chat
Audio, video, chat conferencing
Web App: WebRTC supporting browsers: Google Chrome, Mozilla Firefox, Opera
Mobile Native App: Android, iOS*
- Web socket secure (WSS) connection
- Proprietary communication protocol between server and client
- NAT traversal when using STUN, ICE, TURN*
- Call API
- SIP over UDP (RFC2833)
- SIP Back-to-back UA
- User registration
- Registration pass-through Modus
- DTMF SIP INFO
G.711 A/Ulaw, G722, OPUS, H264, VP8 pass-through
Dynamic jitter control
NAT/NAPT on media
RTP inactivity monitoring
Echo Test service
Dynamical bandwidth estimation and adoption
Active-active redundancy model
Dynamical scaling to fit load requirements
Partitioning (multi domains Support)
Routing by many parameters:
- URI: B-number+Domain
- A-number, source IP, transport protocol, source Domain
Call blocking and filtering
Embedded routing engine
External routing engine
Alternative routing on failure
Secured Web-based UI for configuration and monitoring
Logging of alarms, events, statistics
Troubleshooting via UI
WSS, RTP, SRTP, DTLS, RTCP, SIP UDP,
RFC 4585, RFC 3550, RFC 5104
Translation between transport protocols
Runs on all virtualisation platforms
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