- From: Philipp Hancke via GitHub <sysbot+gh@w3.org>
- Date: Fri, 13 Dec 2019 17:45:08 +0000
- To: public-webrtc-logs@w3.org
> The RTCTransportStats object's bytesSent is said to be: very odd. I'd expect this to be usable to calculate the actual bitrate (w/o ip overhead). This would include any STUN packets sent or received. I *think* that is what Chrome currently does, try this: 1) go to https://webrtc.github.io/samples/src/content/peerconnection/pc1/ 2) make a call, have webrtc-internals open 3) paste pc1.getSenders().forEach(s => s.track.stop()) 4) observe candidate pair stats increase. > So if it's UDP, this would refer to the number of UDP datagrams? Yes. > Same as the number of IP packets? That is rather hard to compare to UDP and may not even be available? If an RTP packet is sent over RTP-over-tcp I'd expect this to increase by 1, no matter how many different ip packets it gets split into. -- GitHub Notification of comment by fippo Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/507#issuecomment-565537538 using your GitHub account
Received on Friday, 13 December 2019 17:45:09 UTC