About Packet Loss

Is there any solution in MediaSoup libraries for Packet Loss? In WebRtcTransport.


  1. TCP
  2. Ping
  3. Restart Ice

Few things that come to mind.

I handle packet loss mostly by allowing clients to detect it and request new ice. If ping times-out I crash the server and leave it out the loop for a little bit.

(Packet loss should be client problem and rare cases server if attacked, if your packet loss is great on the server/machine, I’d not deploy it for other peers.)

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I appreciate your useful and detailed answers.

My work is to repair the Packet Loss, use some QoS strageties,like NACK, retransmission, and so on.

I create 5%, 10%, 30% random packet loss test cases with QNet tools(QNET弱网测试 - 腾讯WeTest),
I hope that the video and audio could still work properly.

I know that enableRtx for PipTransport,

My questions:

  1. Is there a similar solution for WebRtcTransport?
  2. Could it be configured in core\config.js or in src\supportedRtpCapabilities.ts? And how to configure?

Si are you assuming that WebRTC transports don’t have NACK/retransmission capabilities?

I want to make sure WebRtcTransport in MediaSoup libraries have NACK/retransmission capabilities, and how to configure it?
Besides NACK, any other solution in MediaSoup libraries for Packet Loss?