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Statistics and Metrics in WebRTC

Varun Singh
November 04, 2015
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Statistics and Metrics in WebRTC

WebRTC Meetup Yokohama,
Varun Singh
CEO callstats.io

Varun Singh

November 04, 2015
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Transcript

  1. Performance Monitoring Endpoint Endpoint Middlebox MCU, SFU, TURN, … Monitor

    Monitor Monitor Third party Monitor Operations & Management System Implement RTCP Extension Reports http://tools.ietf.org/html/rfc6792 https://tools.ietf.org/html/draft-ietf-xrblock-rtcweb-rtcp-xr-metrics
  2. HTML APIs • HTMLVideoElement Extensions • http://www.w3.org/TR/2014/CR-media-source-20140717/#htmlvideoelement-extensions • droppedVideoFrames, corruptedVideoFrames,

    totalFrameDelay • WebRTC Statistics API • (work in progress) http://www.w3.org/TR/2015/WD-webrtc-stats-20150206/ • PeerConnection Metrics: RTP, datachannel • Transport Metrics: active and passive candidates • Security cipher suite
  3. Edge Stats (ORTC) Lots of data still set to zero.

    but encouraging to see the data
  4. What to measure? • Network metrics • bits per second,

    RTT, jitter, packet losses, … • Multimedia pipeline metrics • playout delay, frames metrics, … • Quality Models from metrics • Annoyances • resolution/frame rate changes, interface changes, … • failures (NATs, insufficient capacity for carrying media, …) • user feedback 10 CAVEAT: not all WebRTC applications care about the same metrics
  5. Low hanging fruit • Endpoint monitoring (Traffic lights) • colors

    to show connection is active/inactive • off/online • good/bad/ugly
  6. Diagnostics • characterise an impairment? • we start with an

    hypothesis… • implemented as decision trees • observe how often it occurs • within a call, across calls, geographies, … • How to characterise quality of the calls? 13
  7. Disruptions Disruption: loss of connectivity when network interfaces change, low

    available capacity,or high delay The light grey vertical lines show disruption, highlighted by the red bounding boxes.
  8. Disruptions and user behaviour User Behaviour: The user tries to

    correct for the disruption by turning on and off video Zoom in to the events to view disruptions
  9. Annoyances Setup times: waiting for the conference to begin: NAT

    traversal, negotiation, etc. Note: gathering and connectivity checks can happen concurrently
  10. • Analytics for WebRTC • 3 lines of code —>

    5 minutes. 1. include <script> 2. initialize() —> needs registration keys 3. addNewFabric(pc, …) —> call started • optionally: send user events: audio muted/unmuted, video paused/resumed, call terminated, call held. • optionally: reportError() —> call failed to be setup due to SDP or Firewalls. • optionally: associateMstWithUserID() —> bridges using multiple media streams in a single peerconnection may want to correlate mediastream quality across endpoints. • optionally: collect and send user feedback http://www.callstats.io/api/