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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. 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.
New Resources on Our WebsiteWe've updated our website with the following new resources:
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