- From: Philipp Hancke via GitHub <sysbot+gh@w3.org>
- Date: Sat, 15 Mar 2025 18:09:56 +0000
- To: public-webrtc-logs@w3.org
Still an issue. For the sake of simplicity try this on https://webrtc.github.io/samples/src/content/peerconnection/audio/ At some point in time I got the following numbers for packetsSent and bytesSent: * outbound-rtp: 2756 packets sent with 136312 bytes sent, 78456 headerBytesSent * transport: 2770 packets sent with 243887 bytes sent * candidate-pair: 2770 packets sent with 243887 bytes sent, 24 requestsSent and 24 responsesSent The difference between outbound-rtp and transport is 16 packets. Too much for DTLS (2). This might be RTCP (14 packets in 55 seconds of audio seems reasonable) which I would expect to be included in outbound-rtp but the spec [disagrees](https://w3c.github.io/webrtc-stats/#dom-rtcsentrtpstreamstats) - how do I learn how many RTCP packets were se nt and what their overhead is? Dividing `243887 - (136312+74568)` by the number of packets sent matches the SRTP overhead (10 bytes for RTP, 14 for RTCP) roughly. May include DTLS. Transport and candidate-pair stats match which is actually what libWebRTC does. I can not be including the STUN requests and responses (48 total) since 2756 + 48 > 2770. -- GitHub Notification of comment by fippo Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/507#issuecomment-2726903133 using your GitHub account -- Sent via github-notify-ml as configured in https://github.com/w3c/github-notify-ml-config
Received on Saturday, 15 March 2025 18:09:57 UTC