Connect WebRTC or RTMP to traditional SIP telephony systems or devices.
SIP Connect feature is designed to make traditional calls across web or mobile phones. It enables interoperability between WebRTC endpoints and existing telephony systems, like your current PBX or phone lines. For instance, a Call Center can send and receive SIP-based calls, using SPLIT function as well.
SIP Connect functionality allows you to send Dual-tone multi-frequency signalling (DTMF) or in-band telecommunication signalling system using the voice-frequency band over SIP. SIP Connect makes it possible to embed advanced web phones or mobile apps into existing infrastructure, making WebRTC easier than ever.
- Send or receive SIP calls and make trunks with external voip systems
- Enable sending DTMF during a voice or video call
- Power any Call Center integration
- Power any IP PBX integration
SIP Connect feature requires Video RTC (WebRTC or RTMP) Gateway On Premise or Cloud CPaaS.
Web & Mobile
SIP Connect is available both for Web and Mobile endpoints for WebRTC. You can seamlessly integrate access from websites and mobile apps into your existing systems.
You can create dual-interfaces with our SDK without changing your business logic.
SIP Connect feature is powered by specific functions inside our API Framework (VideoRTC.js).
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