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JSCommSettings = {
// Web server environment
webserver: {
url_prefix: null // If set, prefix used to construct sound/ URLs
},
// STUN/TURN media relays
stun_servers: [],
turn_servers: [
{ server:"turn:turn1.example.org?transport=udp", username:"bob", password:"bob" }
],
// WebSocket connection
websocket: {
servers: 'wss://ws1.example.org',
connection_recovery_min_interval: 2,
connection_recovery_max_interval: 30
},
// User identity
// If uri == null or sip_auth_password == null the user will be
// greeted with the login box
// If sip_auth_user == null, it will be automatically derived from
// the value of uri
user: {
display_name: 'Bob',
uri: 'sip:bob@example.org',
sip_auth_user: 'bob',
sip_auth_password: 'bob',
sip_auth_user_full_uri: true // Whether to use full URI or just
// the user part to construct the auth user
},
// User SIP registration
registration : {
on_startup: true, // Register when websocket link starts?
expiry: 3600, // Registration expiry (seconds)
user_control: false, // Show button for user to (de)register
server: null // Registration server
},
dialing : {
auto_dial : {
default_destination : 'alice', // A name or full URI
on_startup: false, // Whether to wait for user to click dial
use_video: true // Whether to auto-dial with video
},
edit_destination: true, // Whether the user may edit the destination
show_destination: true, // Whether the user can see the destination
audio_dialing: true, // Whether to offer an audio call button
video_dialing: true, // Whether to offer a video call button
prefer_video: false, // If user presses Enter to dial, use video?
no_answer_timeout: 60, // How long to wait for answer (seconds)
clear_dialbox: false // Whether to clear dialbox after call/attempted call
},
chat : {
enable: true // Whether to show chat functions
},
session : {
show_dtmf_pad : true, // Whether to show a DTMF dialpad during calls
dtmf_duration : 160, // Duration (ms) to signal a DTMF press
dialpad_tone: true, // Whether to play sound after pressing dialpad button
message_tone: true, // Whether to alert (sound) when new IM is received
require_relay_candidate: false // Whether to abort calls if no
// relay candidate found in the SDP
// generated by the browser.
// If a TURN server had been configured,
// the lack of a relay candidate is
// likely to be a sign that UDP is blocked
},
i18n : {
translate: true, // enables the call to i18n init function from JSCommManager.init()
show_menu: true, // show the language chooser menu
default_lang: 'en' // default language to use
},
extra_headers : [
// 'X-WS-Session-Extra: some_token=value'
]
};
/****************************************************************************
JSCommunicator
http://jscommunicator.org
Copyright (C) 2013 Daniel Pocock http://danielpocock.com
Copyright (C) 2014 Juliana Louback http://julianalouback.com
The JavaScript code in this page is free software: you can
redistribute it and/or modify it under the terms of the GNU
General Public License (GNU GPL) as published by the Free Software
Foundation, either version 2 of the License, or (at your option)
any later version. The code is distributed WITHOUT ANY WARRANTY;
without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE. See the GNU GPL for more details.
You may distribute non-source (e.g., minimized or compacted) forms of
that code without the full copy of the GNU GPL normally required
provided you include this license notice and a URL
through which recipients can access the Corresponding Source.
****************************************************************************/
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