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WebRTC publisher adaptive bitrate has high packets lost on both TCP and UDP, Solution?

I try to make publisher perform adaptive bitrate.

The problem is that the video packet loss is very high. I use the provided publish and play project. I remove the enhanceSDP Line. I tried all h264, vp8, vp9. I tried all h264 baseline and main profile level. I tried enable Jitter, sortPacket in my application. Yet, no success.

I tried with 500, 1000, 2000 kbps upload speed. the result is similar.

Only scenario that works well is that max bitrate sent must be less or equal to the publisher upload speed.

Do you have any suggestion or solution in this case?

wowza_access.log

o=WowzaStreamingEngine-next 1217336901 2 IN IP4 127.0.0.1

s=-

t=0 0

a=group:BUNDLE audio video

a=msid-semantic: WMS *

m=audio 9 UDP/TLS/RTP/SAVPF 111

c=IN IP4 0.0.0.0

a=rtcp:9 IN IP4 0.0.0.0

a=ice-ufrag:71e69b13

a=ice-pwd:e9019472363a3c41d4ab5fc0731ff777

a=ice-options:trickle

a=fingerprint:sha-256 D1:8F:02:1F:FF:7C:B0:EB:61:AD:29:04:B5:27:13:97:47:C8:CB:DD:69:A7:88:D5:F1:4D:1F:44:BB:08:0D:37

a=setup:passive

a=mid:audio

b=CT:48

b=AS:48

a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level

a=recvonly

a=rtcp-mux

a=rtpmap:111 opus/48000/2

a=fmtp:111 x-google-min-bitrate=6;x-google-max-bitrate=48

a=rtcp-fb:111 transport-cc

a=fmtp:111 minptime=10;useinbandfec=1

m=video 9 UDP/TLS/RTP/SAVPF 100

c=IN IP4 0.0.0.0

a=rtcp:9 IN IP4 0.0.0.0

a=ice-ufrag:71e69b13

a=ice-pwd:e9019472363a3c41d4ab5fc0731ff777

a=ice-options:trickle

a=fingerprint:sha-256 D1:8F:02:1F:FF:7C:B0:EB:61:AD:29:04:B5:27:13:97:47:C8:CB:DD:69:A7:88:D5:F1:4D:1F:44:BB:08:0D:37

a=setup:passive

a=mid:video

b=CT:2500

b=AS:2500

a=framerate:25

a=extmap:2 urn:ietf:params:rtp-hdrext:toffset

a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

a=extmap:4 urn:3gpp:video-orientation

a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01

a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay

a=recvonly

a=rtcp-mux

a=rtcp-rsize

a=rtpmap:100 H264/90000

a=fmtp:100 x-google-min-bitrate=100;x-google-max-bitrate=2500

a=rtcp-fb:100 ccm fir

a=rtcp-fb:100 nack

a=rtcp-fb:100 nack pli

a=rtcp-fb:100 goog-remb

a=rtcp-fb:100 transport-cc

a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e014

webrtc logs