I don’t understand why I have 2 sec latency. I try to find some utilities like in xsplit broadcaster. But I didn’t find it. And by the way, I used xsplit broadcaster test bandwidth. I tried to connect and it showed me result from 1046 ms to 2078 ms. Of course, when I stream on server I have 1-2 sec delay. I saw the streams in web players (with netbuffer 0.1) and in native players from linux, mac and windows (vlc). I saw all articles on forum which describe latency. I tried this https://www.wowza.com/resources/4.0.0/examples/VideoChat/FlashRTMPPlayer/player.html and from this player I have 0.8-1 sec latency. But it’s not good( For ex. when we test polycom we saw latency about 50 ms!
And my question is what params in config I need to fix to minimaze this latency?
I spoke about rtmp, rtsp protocols. But it actual and for Ndvr, hls (native, without ndvr plugin)
The increase in latency might be also caused by the bad internet connection between the video chat client (encoder) and Wowza server to which the encoded stream is being pushed. If the internet bandwidth is too low to support the bitrate of the stream being pushed to Wowza, then the server will start buffering internally the incoming stream in order to properly construct the frames that are going to be pushed to the client playing back the stream.
Can you perform a test by lowering the bitrate of the encoded stream from the video chat application so that the the total bitrate can be supported by the internet bandwidth available between the video chat client and Wowza.
Hello there, I am a little confused as to what exactly you are doing. But 1 - 2 seconds is pretty good for HLS.
HTTP streaming sends chunks to the client. The client needs 3 chunks cached before it starts playing.
By default Wowza is set to send 3, 10 second chunks in each packet sent to the client. You can modify this behavior by editing the Application.xml file.
This guide explains how to control how the Cupertino (iOS device) segmenter segments an incoming live stream:
Hello there, I am a little confused as to what exactly you are doing. But 1 - 2 seconds is pretty good for HLS.
HTTP streaming sends chunks to the client. The client needs 3 chunks cached before it starts playing.
By default Wowza is set to send 3, 10 second chunks in each packet sent to the client. You can modify this behavior by editing the Application.xml file.
This guide explains how to control how the Cupertino (iOS device) segmenter segments an incoming live stream:
Chunks must start on a key frame. So it is best to use a key frame interval that is factor of the cupertinoChunkDurationTarget setting.
Try 2 second key frame frequency and cupertinoChunkDurationTarget “2000” (2 seconds)
I hope this helps.
Kind regards,
Salvadore
Salvadore, thank you for your help! But you didn’t understand me fully. I spoke about latency in rtmp (rtsp) protocol
In hls I have 13 sec latency. For hls I wound be perfect if latency will be 2 sec. But for rtmp, as i understand, it’s not good. It’s if we speak about video chat between 2 people.
I try videochat application. And latency was less then 1 sec. But after 30 sec for example, I saw that latency accumulate and final latency is 2 sec, after 5 min, it will be more and more. And I couldn’t understand why this thing is.