Voice freezing after some seconds


I'm using Photon Voice and everything works great but I'm experiencing an issue that people start to talk and everything is fine and then it just stops and will continue after some seconds again. The server is based in the EU but the guys I was trying this are based in the US. Is this caused by network lags? Should I decrease Bitrate, Sample Rate or what can I do about it? I really think it is annoying and should not happen at all.

Looking forward for some information what I could do about this :)


  • vadimvadim ✭✭✭

    Does your app log something when audio stops and starts playing back again? You may need to enable more logging with PhotonVoiceSettings.Instance.DebugInfo = true. How many clients are in the room? What if only one client is joined in DebugEcho mode?
    Does it plays back from the point where it stopped or just drops chunk of data?

    Try to increase PlayDelayMs parameter which sets the size of jitter buffer in terms of milliseconds.
    You can decrease Bitrate to send less data. This will help if client exceeds the bandwidth.

    To diagnose your connection, run TestVoice-Scene and look at "RTT/Var/Que" "Data rate i/out" and "Frames lost" numbers.
  • DummieDummie
    edited October 2016
    Thank you for your answer and sorry for my delay in response. I can only do some further testing at the weekends.

    I'm seeing lots of warnings and messages like that: http://img4host.net/upload/24110758580dcf6ec7cd5.png

    It slows the game down a lot and I'm not sure if that should happen? Often I also see some error that there are two instances of PUN Voice. Not sure if that should happen.

    However the issue still remains and I already set the buffer to 300 ms. Should I go even higher like 500 ms? And what is Frame Duration (set to 20 ms) about? I could not find any information about it. I'm aiming for 90 fps because it becomes a VR game. Should I apply changes here as well?
  • vadimvadim ✭✭✭
    If it helps, you can try bigger buffer size for at least to find out the reason for the issue.
    Frame duration means interval in which audio data packets are sent. Lower values reduce latency but increase data overhead. Default value should work for most cases.
    Did you get the numbers I asked for in previous post?
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