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Bug 970685 - definitions for WebRTC telemetry for jitter, packet-loss and RTT. r=smaug, r=jesup

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This commit is contained in:
Jan-Ivar Bruaroey 2014-05-12 15:35:34 -04:00
parent 40142335ba
commit 07463bc134
2 changed files with 70 additions and 1 deletions
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1
dom/webidl/RTCStatsReport.webidl
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@ -131,7 +131,6 @@ callback RTCStatsReportCallback = void (RTCStatsReport obj);
dictionary RTCStatsReportInternal {
DOMString pcid = "";
sequence<RTCRTPStreamStats> rtpStreamStats;
sequence<RTCInboundRTPStreamStats> inboundRTPStreamStats;
sequence<RTCOutboundRTPStreamStats> outboundRTPStreamStats;
sequence<RTCMediaStreamTrackStats> mediaStreamTrackStats;

70
toolkit/components/telemetry/Histograms.json
View File

@ -5152,6 +5152,76 @@
"n_buckets": 1000,
"description": "The delay (in milliseconds) when video is behind audio. Zero delay is not counted. Measured every second of a call."
},
"WEBRTC_VIDEO_QUALITY_INBOUND_PACKETLOSS": {
"expires_in_version": "never",
"kind": "exponential",
"high": 1000,
"n_buckets": 100,
"description": "Locally measured packet loss on inbound video (permille). Sampled every second of a call."
},
"WEBRTC_AUDIO_QUALITY_INBOUND_PACKETLOSS": {
"expires_in_version": "never",
"kind": "exponential",
"high": 1000,
"n_buckets": 100,
"description": "Locally measured packet loss on inbound audio (permille). Sampled every second of a call."
},
"WEBRTC_VIDEO_QUALITY_OUTBOUND_PACKETLOSS": {
"expires_in_version": "never",
"kind": "exponential",
"high": 1000,
"n_buckets": 100,
"description": "RTCP-reported packet loss by remote recipient of outbound video (permille). Sampled every second of a call (for easy comparison)."
},
"WEBRTC_AUDIO_QUALITY_OUTBOUND_PACKETLOSS": {
"expires_in_version": "never",
"kind": "exponential",
"high": 1000,
"n_buckets": 100,
"description": "RTCP-reported packet loss by remote recipient of outbound audio (permille). Sampled every second of a call (for easy comparison)."
},
"WEBRTC_VIDEO_QUALITY_INBOUND_JITTER": {
"expires_in_version": "never",
"kind": "exponential",
"high": 10000,
"n_buckets": 100,
"description": "Locally measured jitter on inbound video (ms). Sampled every second of a call."
},
"WEBRTC_AUDIO_QUALITY_INBOUND_JITTER": {
"expires_in_version": "never",
"kind": "exponential",
"high": 10000,
"n_buckets": 1000,
"description": "Locally measured jitter on inbound audio (ms). Sampled every second of a call."
},
"WEBRTC_VIDEO_QUALITY_OUTBOUND_JITTER": {
"expires_in_version": "never",
"kind": "exponential",
"high": 10000,
"n_buckets": 1000,
"description": "RTCP-reported jitter by remote recipient of outbound video (ms). Sampled every second of a call (for easy comparison)."
},
"WEBRTC_AUDIO_QUALITY_OUTBOUND_JITTER": {
"expires_in_version": "never",
"kind": "exponential",
"high": 10000,
"n_buckets": 1000,
"description": "RTCP-reported jitter by remote recipient of outbound audio (ms). Sampled every second of a call (for easy comparison)."
},
"WEBRTC_VIDEO_QUALITY_OUTBOUND_RTT": {
"expires_in_version": "never",
"kind": "exponential",
"high": 10000,
"n_buckets": 1000,
"description": "Roundtrip time of outbound video (ms). Sampled every second of a call."
},
"WEBRTC_AUDIO_QUALITY_OUTBOUND_RTT": {
"expires_in_version": "never",
"kind": "exponential",
"high": 10000,
"n_buckets": 1000,
"description": "Roundtrip time of outbound audio (ms). Sampled every second of a call."
},
"WEBRTC_CALL_DURATION": {
"expires_in_version": "never",
"kind": "exponential",