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.

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Install the LiveSwitch gateway and media servers in your infrastructure and start building your own application now. No credit card required.

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What is LiveSwitch?

Combining peer-to-peer audio/video with server-based selective forwarding and mixing.

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