[webrtc-stats] Lifetime of "outbound-rtp" (#558)

henbos has just created a new issue for https://github.com/w3c/webrtc-stats:

== Lifetime of "outbound-rtp" ==
The type definition for ["outbound-rtp"](https://w3c.github.io/webrtc-stats/#dom-rtcstatstype-outbound-rtp) says:
Statistics for an outbound <a>RTP stream</a> that is currently sent with this
{{RTCPeerConnection}} object.

But [The RTP statistics hierarchy](https://w3c.github.io/webrtc-stats/#the-rtp-statistics-hierarchy) says:
RTP monitored objects are not deleted.

The intent being not to delete "outbound-rtp" stats objects, the first paragraph should say "that is currently sent or has ever been sent by this RTCPeerConnection". This is a simple PR.

But note that the modern getStats() implementation in chromium does not implement this. And if it did...
- How can you tell which RTP streams are active and which are old? Do we need a new stat?
- What happens to trackId, senderId, mediaSourceId and transportId of an inactive rtp stream?

Alternatively, we could make the spec align with current implementations and say that they are deleted when the sender is reconfiured to send a different ssrc or if the sender stops sending.

Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/558 using your GitHub account

Received on Tuesday, 21 April 2020 07:45:32 UTC