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WebRTC is a free, open project that enables web browsers with Real-Time Communications (RTC) capabilities via simple JavaScript APIs. The WebRTC components have been optimized to best serve this purpose.Pricing:
- Open Source
In most cases, these video communication applications use WebRTC as their desired communication protocol. This protocol splits each media type into separate tracks, for example, when a participant joins the call, it’s going to have at least two corresponding tracks – one for audio, one for video. When screen-sharing is turned on, two more tracks are created – one for the audio, the other for the video. The newly created tracks aren’t any different from what is regularly made when participants enable a webcam. This means that sharing a screen is almost identical to a new participant joining the call. This allows gathering some metrics that could indicate whether the feature works, and quality was good and stable. Loadero is used a lot for testing audio and video quality in apps that use webcams and microphones, and a similar approach to WebRTC testing can be used to test screen sharing as well.
#Video Conferencing #Group Chat & Notifications #Communication 46 social mentions
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Join a WebRTC video conference powered by the Jitsi Videobridge
(client) => { const callDuration = 120 * 1000; const timeout = 30 * 1000; client .url(`https://meet.jit.si/one-on-one-${client.globals.run.id}`) .waitForElementVisible(".field", timeout) .setValue(".field", `R${client.globals.run.id}_P${client.globals.participant.id}`) .waitForElementVisible('[aria-label="Join meeting"]', 10 * 1000) .pause(2 * 1000) .click('[aria-label="Join meeting"]') .perform(() => { if (client.globals.group.id === 1 && client.globals.participant.id === 0) { client.click('[aria-label="Start / Stop sharing your screen"]') } }) .pause(callDuration / 2) .takeScreenshot("WebRTCCall.png") .pause(callDuration / 2); };.
#Communication #Group Chat & Notifications #Messaging 141 social mentions