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Cost Optimized Migration from Lagacy
to Voip Technologies

Winner of ‘Real Time Web Solutions
Excellence Award’

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.
  • 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.

Features

  • 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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