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mikebrown  
#1 Posted : Friday, May 22, 2020 3:21:47 AM(UTC)
mikebrown

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Hi All!

I've read the previous posts about the jitter buffer. I had a rehearsal with 2 remote presenters where the latency would start low 50-120ms. Then slowly build to 500-600ms, then it would jump the latency back down and I would hear a pop and miss a word or two. This process would repeat throughout the test.

The vMix computers internet connection is better than 500 mbs up/500 mbs down, and the quality of the presenters stayed at 1280x720 so it seems to be purely a latency issue.

Any suggestions?
ask  
#2 Posted : Friday, May 22, 2020 9:39:07 AM(UTC)
ask

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Originally Posted by: mikebrown Go to Quoted Post
Hi All!

I've read the previous posts about the jitter buffer. I had a rehearsal with 2 remote presenters where the latency would start low 50-120ms. Then slowly build to 500-600ms, then it would jump the latency back down and I would hear a pop and miss a word or two. This process would repeat throughout the test.

The vMix computers internet connection is better than 500 mbs up/500 mbs down, and the quality of the presenters stayed at 1280x720 so it seems to be purely a latency issue.

Any suggestions?


OK. This has been said many, many times. Your connection is only part of the pipe through the internet. Speed tests on your connection only test between your PC and the nearest test server, not you and your guest. Any particular segment in the internet pipe that the call uses could be at issue. WebRTC is designed to auto-negotiate that connection and will dynamically adjust itself to the conditions. This is not something that vMix can control. The only thing you can do is to drop the call, reconnect and establish a new path for the connection.
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