WebRTC to SIP Gateway

WebRTC to SIP Gateway

WebRTC-SIP gateway is an award winning solution which uses WebRTC technology to receive voice/video calls from any browser or mobile application on your SIP network or end points without downloading any plugins.

WebRTC-SIP Gateway (Overview)

Works as a mediator between two types of VOIP transport mediums.
Converts SIP over websockets to SIP over UDP and encrypted RTP over DTLS (Secure UDP) to plain RTP over UDP.
Enables user to make VOIP calls originate from browser and terminate on conventional SIP switches.

How does it Work?

• Client application uses Token_generator file to generate authentication token.
• Client application sends this generated token to WebRTC enabled devices (browser or android apps).
• Calls can be initiated from these devices using JavaScript API provided to specified SIP switch phones or PSTN phones.


• Robust and Performant Gateway.
• Scalability as per need.
• Flexible, plug & play module interface.
• Highly secure communication and easily managed.
• Call, Media & Codec control