Comments (3)
Copying my comment from the issue #68:
With the frozen timestamps, an app can filter out the older candidates, assuming the app has some idea about how often it queries or when it last queried the stats.
Although one concern I do have (and perhaps could be solved in a different way) is if the ice candidate is currently inactive and was active in the past, we do not really know when it was last active. We perhaps need a lastPacketSentTimestamp, lastPacketReceivedTimestamp or similar
from webrtc-stats.
Would this timestamp be valid for only local candidates, or would we also update the remote candidate of a pair? Or perhaps it's on candidate pairs? (a candidate can't be active without being part of a pair).
from webrtc-stats.
I will add the timestamps to the candidate pair stats.
from webrtc-stats.
Related Issues (20)
- Conformance criteria incompatible with standards-track HOT 7
- RTX and FEC stats are incomplete. HOT 12
- Should we flatten the hierarchy? HOT 5
- Change playoutdelay to jitterbuffertarget HOT 2
- Add clarification that inbound-rtp.bytesReceived includes retransmissions. HOT 6
- RTCAudioSourceStats.audioLevel - what algorithm described in totalAudioEnergy? HOT 3
- Delete the obsolete section, "track" and "stream" stats - they don't exist anymore HOT 3
- Improve test coverage HOT 1
- add framesInput stats to rtcoutboundrtpstreamstats HOT 6
- Stats example uses confusing variable names
- Exposing audio interruption metrics to JavaScript
- Define a mechanism for setting the trigger duration for a video freeze
- Sender-side packetsReceived can be negative from garbage RRs HOT 1
- Change "Does not" to "must not"
- Make "not present" reference "map/exist"
- Is codec.sdpFmtpLine present when there is no fmtp line in the SDP? HOT 2
- Lifetime of outbound-rtp should start BEFORE first packet is sent HOT 3
- Should RTCRtptransceiver.stop() cause inbound-rtp stats to disappear?
- RTCCodecStats.clockRate - media sampling rate or the codec clock rate?
- RTCStats.timestamp - fingerprinting and since epoch
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from webrtc-stats.