WebRTC SIP Integration 
to Support VoIP/PSTN Architectures

Integrating Legacy Telephony into WebRTC Audio/Video Calls

LiveSwitch allows for the easy integration of VoIP and PSTN into a LiveSwitch WebRTC conference.



Full SIP Integration

LiveSwitch makes it easy to add voice-only VoIP/PSTN clients into your WebRTC conference calls.

The SIP Connector service included with LiveSwitch lets you register your application with a SIP server (e.g. a SIP trunk or PBX) for the purposes of accepting incoming calls. Calls can be routed to a specific channel/session based on the inbound number, or you can add in a virtual PBX to intercept the call and prompt for an access ID.

The inbound SIP audio stream is automatically routed to the LiveSwitch media server for forwarding/mixing to the other participants, while the outbound SIP audio stream is fed with a mix of audio from all the other participants.

Ready to Try LiveSwitch?

Pricing that scales up when you do.

Buy LiveSwitch Now


Download the Whitepaper

Looking for more information on LiveSwitch? Download our whitepaper now.

Download Now

The Ideal Solution for Many Industries

Eliminate High Usage Costs
with One-Time Licensing


See All Pricing