I’m using Wowza Stream Engine running on AWS to provide WebRTC streams from a SRT source. We are getting Jerky playback with the WebRTC playback on certain devices. If I look at it as MPEG DASH the live source plays fine, so i’m confident the source is ok, but when viewed as WebRTC its occasionally good and smooth, but more often than not Jerky.
Looking at the WebRTC internals (within Chrome) we do show a fair bit of packet loss, and the NACK count is quite high.
Does anyone have any ideas how to help smooth out the WebRTC video? It is in use for a Live TV production monitoring application, so Jerky video isn’t ideal, but low latency is required.
Incoming Streams are H264, around 2Mbs, 1280x720, 25fps, and other than changing the audio to OPUS, I’m doing no other transcoding on the Wowza server.