Is there any solution in MediaSoup libraries for Packet Loss? In WebRtcTransport.
- 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.)
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,
- Is there a similar solution for WebRtcTransport?
- 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?