I have cameras connected in the local network and I'm using Wowza server & WebRTC to be able to play those streams in the clients using video element without ActiveX plugins. In the client I'm working with Websocket to get the data from Wowza and passing it to the RTCPeerConnection and to the video tag.
I've made everything work. so no problem there. The stream is playing well with low latency (at least in the beginning).
My question is:
At the beginning of the play the latency is very low less than a second - like needed. After few seconds the latency grows and keep on growing.. I guess because of our Wowza server is weak. Am i right ?
If so, Is there a way to configure it in the client so it will drop old buffered chunks and play the last ones ? so i will always be as much as live as possible regardless the server strength/Bandwidth/... ? I can understand that i will loose the smoothness of the video, and there will be jumps.
Did you look at webrtc-internals ? did you see any package drop or jitter etc ? you can also check the peer statistic and get this info per second, when you refresh the page, if you watch the stream with low latency, so there is no server side issue, but still high latency, you can look at the stream key frame and bandwidth, the webrtc suggested bitrate about 2.5 mbit ( highest )