REVE WebRTC – SIP Gateway

Winner of Real Time Web Solutions Excellence Award

REVE 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.

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Reve 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.
On browser side it uses WebRTC technology to transfer media, which is supported by most of the popular browsers
Doesn't require any kind of third party plugins
Uses standard SIP for signaling

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

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