WebRTC Broadcasting 
for All Browsers, Mobile and Native Platforms

Broadcasting and Recording WebRTC-based Audio/Video Calls

Many applications are designed around a large number of people viewing a limited number of audio and/or video feeds, often just one or two.

LiveSwitch’s internal forwarding architecture makes this easy, and can scale out to extremely high subscriber counts while maintaining sub-second latency.

 

LiveSwitch allows you to transmit audio, video, and data streams directly to an audience consisting of any number number of viewers.

In cases with a single broadcaster, viewers can connect in with lowest latency by using LiveSwitch’s SFU capability, which bypasses media decoding on the server and forwards packets directly to the viewers.

If multiple broadcasters are involved, LiveSwitch’s MCU can be used to deliver the combined feed as a single stream, at a cost of server resources to decode, mix, and re-encode the broadcasters’ media feeds.

LiveSwitch allows real-time text-based communications such as chat to take place while the conference is active. This can be used to support any number of custom scenarios for real-time interaction, such as allowing a viewer to “raise their hand” to the broadcasters who can then temporarily allow the new participant to contribute to the live audio and/or video feed.

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What is LiveSwitch?

Combining peer-to-peer audio/video with server-based selective forwarding and mixing.

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LiveSwitch Features SIP Integration

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