I would like to ask for help that how can I trace for the webrtc rtp connection has no data packets sometimes.

It’s not 100% reproducible, but sometimes make a call has no early media ring back tone and also accept call cannot hear both side voice.

when the issue occured, the peerConnection?.getStats() will be as below:

rtc inbound-rtp: {"timestamp":1715323975903,"type":"inbound-rtp","kind":"audio","jitter":0,"packetsLost":0,"packetsReceived":0,"bytesReceived":0,"headerBytesReceived":0,"jitterBufferDelay":0,"totalAudioEnergy":0,"totalSamplesReceived":0,"totalSamplesDuration":0}
rtc outbound-rtp: {"timestamp":1715323975903,"type":"outbound-rtp","kind":"audio","packetsSent":0,"retransmittedPacketsSent":0,"bytesSent":0,"headerBytesSent":0,"retransmittedBytesSent":0}

It seems that rtp connection not established?

sip.js acquire local media stream doesn’t show any error and not see any microphone access permission issue. After the issue happened, then end the call and make a new call, then everything will be good again.

Any idea or any suggestion is welcomed.
Thanks a lot.