WebRTC high latency & sound issues

[Bohdan_Klochko]

I am using Wowza Streaming Cloud trial for creating WebRTC broadcasts, but I am currently facing 2 issues:

1. The delay between broadcasting & playback is around 40 seconds, but as I understand WebRTC protocol allows you to run almost real-time broadcasts (1-2s delay).
2. The sound from the broadcast loops and plays a few times in playback.

Is there any way to fix these issues?

[Rich_Millard]

I’m also curious about the same issue. I was seeing latency closer to 12-15 seconds when viewing the stream embedded on my personal website. I also need the stream to be as close to real time as possible. Sub 2 seconds. I’m using a Teradek VidiuGo to encode the stream.

[Rose_Power-Wowza_Com]

You need to select low latency streaming in Wowza Cloud manager when you set up a new stream. The defaults are around 30 seconds but if you click “Yes, create an HLS low latency stream” it’ll get the chunk size down to about 6 seconds latency.

The Streaming Cloud workflow is WebRTC ingest and we transmux it for HLS playback. For one second streaming, you are referring to WebRTC ingest and WebRTC playback which is available in Streaming Engine, not Cloud. Cloud is not WebRTC playback.

[Rose_Power-Wowza_Com]

Also, make sure you have Adaptive bitrate selected. For Live Stream Type, select Adaptive bitrate. Passthrough audio streams aren’t available for WebRTC in Cloud.

[Bohdan_Klochko]

Thanks for your help, Rose. It really helps!

[Amara_Graham]

Rich, did you follow Rose’s guidance below and see difference?