Server-Based Forwarding and Mixing - Coming June 2017

Scaling is expensive and difficult... Unless you use LiveSwitch.

LiveSwitch allows you to seamlessly choose between a direct "peer-to-peer" mode, a "forwarded" media mode and a "mixed" media mode. You can even utilize all three types of clients in the same session! This lets you maximize scalability and minimize costs by only processing the absolute minimum amount of data required for your use case.

True to WebRTC spirit, it will also be signaling agnostic. The included gateway can be extended to support third-party signaling systems as well as VoIP, PSTN, and content delivery networks.

When you purchase LiveSwitch, you'll also receive our multi-award winning IceLink, the industry leader in flexible P2P audio/video streaming for WebRTC.

Click for More information on LiveSwitch

How Can I Use SIP with WebRTC?

If you’re asking this question, then chances are you either have an existing SIP infrastructure and are looking for a way to interconnect with WebRTC-enabled endpoints, or you have an existing web application and are looking for a way to interconnect with the telephone network.

In both cases, the goal is the same - interconnection - and that’s what SIP is all about.


Continue Reading on the Blog


New Resources on Our Website

We've updated our website with the following new resources:

  • Newsletters Archive - View copies of previous news releases, product updates and more.
  • White Papers - Over the next few months, we'll be publishing several white papers on WebRTC and iRTC. See them all here.
  • Blog RSS Feed - By request (thanks Tsahi!), we've added an RSS feed to our blog. Subscribe to stay up to date with breaking news.


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