WebRTC-SIP Gateway
From your website to your service team.
Let visitors start a voice call from their browser and reach your existing PBX or contact center, without installing a separate calling application.
What WebRTC-SIP Gateway brings to your team.
TeleFinity WebRTC-SIP Gateway connects browser communication to enterprise SIP infrastructure. Add a calling entry point to your website, keep your existing service destinations and choose cloud or on-premises deployment.
Browser calling
Turn a supported desktop or mobile browser into a customer communication endpoint.
SIP connectivity
Connect browser calls to your existing Call Manager, contact center or traditional PBX through the gateway.
Website integration
Embed the JavaScript calling code supplied with the activated service.
Deployment choice
Use a cloud service or discuss deployment inside your own data center.
DTMF and call records
Capture keypad events and call detail records for the configured service.
Service options
Discuss recording, location restrictions and concurrency requirements for your deployment.
From interaction to outcome.
Website visitor
Browser voice call
WebRTC-SIP gateway
PBX or contact center
A few things you may be wondering.
Get clarity before your next step.
Do callers need to install software?
No separate calling application is required for supported browsers. The caller needs a microphone, permission to use it, and a suitable internet connection.
Do we need to replace our PBX?
The gateway is designed to connect browser calls to existing voice destinations. SIP connectivity and compatibility are checked during deployment planning.
Continue the experience.
TeleFinity Contact Center
Contact center software
Explore TeleFinity Contact CenterActFinity
IVR and voice automation
Explore ActFinitySee WebRTC-SIP Gateway in your world.
Walk through your service scenario, integration needs and deployment requirements with TeleFinity.