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Jitsi rtcstats client\nRtcstats client [fork](https://github.com/fippo/rtcstats) tailored for jitsi-meet integration. Server repository can be found here [rtcstats-servers](https://github.com/jitsi/rtcstats-server).\n\n## Description\nThe rtcstats ecosystem consists of a javascript client library which sends statistics and a node.js server which gathers and processes them.\n\nThis repo represents the client side component. It's meant to run in a browsers/electron environment which exposes GUM and WebRTC standard functionality.\n\nIn short, once integrated, the library overwrites `GUM` and `RTCPeerConnection` functionality and proxies most calls and events going through them, sending the gathered data via a websocket to the rtcstats-server. On top of that, each newly created RTCPeerConnection has a configured interval set for it, which calls getStats periodically, this too is sent to the rtcstats-server.\n\n## Installation\nThe project is organised as simple ES6 modules that can be easily imported into jitsi-meet. Originally rtcstats had a more generic approach and was bundled for maximum compatibility, this created issues when importing into jitsi-meet so that option is no longer supported, it now relies on jitsi-meet to do bundling and transpiling.\n\nTo install simply.\n```\nnpm install github:jitsi/rtcstats#vx.x.x\n```\n\n## Usage\nIn order to initialise rtcstats the following steps are required:\n```javascript\nimport rtcstatsInit from 'rtcstats/rtcstats';\nimport traceInit from 'rtcstats/trace-ws';\n\n/**\n * Initialises the trace object which is the channel that rtcstats uses to send data.\n *\n * @rtcstatsEndpoint - rtcstats-server endpoint ex: \"wss:\\\\sample-rtcstata-endpoing.org:3000\"\n * @handleTraceWSClose - callback for handling websocket closed event.\n */\nconst trace = traceInit(rtcstatsEndpoint, handleTraceWSClose);\n\n/**\n * Initialises rtcstats, overwrites GUM and RTCPeerConnection and starts sending data.\n *\n * @trace - trace channel on which data is sent.\n * @pollInterval - interval at which getStats is called and sent.\n * @prefixesToWrap - legacy RTCPeerConnection prefixes for older browser compatibility. Almost all browser now support the RTCPeerConnection API so it can be left empty\n * @connectionFilter - callback used to filter out RTCPeerConnections based on their config.\n */\nrtcstatsInit(trace, pollInterval, ['', 'webkit', 'moz'], connectionFilter);\n```\nBecause `GUM` and `RTCPeerConnection` are overwritten, rtcstats needs to be initialized before any aliases to them are created. For instance lib-jitsi-meet doesn't directly call these functions but rather has references, thus initializing rtcstats after lib-jitsi-meet would result in the original methods being called and\nnot those that are proxied.\n\nIf you need things like a client or conference identifier to be sent along, the recommended way is to use the legacy peerconnection constraints when constructing your RTCPeerConnection like this:\n\n```javascript\nvar pc = new RTCPeerConnection(yourConfiguration, {\n  optional: [\n    {rtcStatsClientId: \"your client identifier\"},\n    {rtcStatsPeerId: \"identifier for the current peer\"},\n    {rtcStatsConferenceId: \"identifier for the conference, e.g. room name\"}\n  ]\n})\n```\n\nIf that integration is not possible there is a fallback integration which allows\nsending per-client information about the user id and conference id. This\ncan be used by calling\n```javascript\ntrace('identity', null, {user: 'your client identifier',\n    conference:'identifier for the conference, e.g. room name'});\n```\n\nWhen using ontop of adapter it is typically not necessary (and potentially harmful) to shim the webkit and moz prefixes in addition to the unprefixed version.\n\n## Details\nThe client overwrites and proxies the following functions and associated events:\n* getUserMedia, getDisplayMedia. Data such as parameters which gum used and the outcome of the operation is sent to the server.\n* RTCPeerConnection.\n  Constructor parameters are sent to the server.\n  By having control over the c’tor  the client adds listeners to several events of interest on a newly created peer connection object, such as.\n  * icecandidate\n  * addstream\n  * track\n  * removestream\n  * signalingstatechange\n  * iceconnectionstatechange\n  * icegatheringstatechange\n  * connectionstatechange\n  * negotiationneeded\n  * datachannel\n\nData regarding each event is sent to the server.\n\nRTCPeerConnection methods are also hooked into and parameters sent to the server:\n* createDataChannel\n* addStream, removeStream\n* addTrack\n* removeTrack\n* addTransceiver\n* createOffer\n* createAnswer\n* setLocalDescription\n* setRemoteDescription\n* addIceCandidate\n\nWhen a participant leaves a conference, the server will have a complete overview of the gum and peer connection flows.\nAt this point the server will begin extracting a “feature set”, which is sent to a database, once this is complete the statistics dump is stored on s3.\nThe s3 dump can be visualized, giving you an almost chrome://webrtc-internals view of the participants sessions see [Importing the dumps](##-Importing-the-dumps).\n\n## Importing the dumps\nThe dumps generated can be imported and visualized using [this tool](https://fippo.github.io/webrtc-dump-importer/rtcstats)\n\n## Authors and acknowledgment\nThe project is a fork of https://github.com/fippo/rtcstats thus proper thanks are in order to the original contributors.","users":{}}