URL: https://linuxfr.org/forums/general-cherche-logiciel/posts/serveur-d-appel-video-jitsi-meet
Title: Serveur d'appel vidéo : jitsi meet
Authors: electro575
Date: 2019年11月17日T21:42:16+01:00
License: CC By-SA
Tags: jitsi
Score: 2
Bonsoir à tous,
Je suis en pleine mise en oeuvre d'un serveur de conférence vidéo.
J'étais interessé par jitsi-meet depuis un petit moment.
Ce soir je le déploie et la je vois en modifiant le code suivant des lignes pour google.
/etc/jitsi/meet/jitsi.example.com-config.js
```
 p2p: {
 // Enables peer to peer mode. When enabled the system will try to
 // establish a direct connection when there are exactly 2 participants
 // in the room. If that succeeds the conference will stop sending data
 // through the JVB and use the peer to peer connection instead. When a
 // 3rd participant joins the conference will be moved back to the JVB
 // connection.
 enabled: false,
 // Use XEP-0215 to fetch STUN and TURN servers.
 // useStunTurn: true,
 // The STUN servers that will be used in the peer to peer connections
 stunServers: [
 { urls: 'stun:stun.l.google.com:19302' },
 { urls: 'stun:stun1.l.google.com:19302' },
 { urls: 'stun:stun2.l.google.com:19302' }
 ],
 // Sets the ICE transport policy for the p2p connection. At the time
 // of this writing the list of possible values are 'all' and 'relay',
 // but that is subject to change in the future. The enum is defined in
 // the WebRTC standard:
 // https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
 // If not set, the effective value is 'all'.
 // iceTransportPolicy: 'all',
 // If set to true, it will prefer to use H.264 for P2P calls (if H.264
 // is supported).
 preferH264: true
 analytics: {
 // The Google Analytics Tracking ID:
 // googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
 // The Amplitude APP Key:
 // amplitudeAPPKey: '<app_key>'
 // Array of script URLs to load as lib-jitsi-meet "analytics handlers".
 // scriptURLs: [
 // "libs/analytics-ga.min.js", // google-analytics
 // "https://example.com/my-custom-analytics.js"
 // ],
```
J'aimerais savoir comment contourner ce genre de code.
Connaitre également ce que vous avez mis en oeuvre comme solution.
J'étais interessé par Ring et puis on m'a proposé linphone aussi, mais passe t-on par son propre serveur maison ?
https://opensource.com/alternatives/skype
Merci à tous
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