WebRTC Client Options (JSON)
Options for WebRTC Client
| Key | Type | Explanation |
| ringTone | boolean
or string |
Set ring tone.
true : ring a default tone. false : disable ringing tone. <URL> : ring the audio located at the specified address by the URL. |
| autoAnswerTone | boolean
or string |
Set an auto answer tone.
true : Enable an auto answer (default answer tone is used). false : Disable auto answer. <URL> : ring the audio located at the specified address by the URL. |
| disconnectTone | boolean
or string |
Set a disconnect tone.
true : Enable a disconnect tone (default tone is used). false : Disable a disconnect tone. <URL> : ring the audio located at the specified address by the URL. |
| dtmfTone | boolean | Set DTMF push tone. The user who pushes DTMFs can hear the tones.
true : Enable push tone. false : Disable push tone. |
| dtmfSendMode | number | Set DTMF sending mode
1: Inband DTMF audio signal is used with priority if it can be used. In other cases, SIP INFO DTMF is used. Also the DTMF sending mode can be changed by the “Widget.setDtmfSendMode(number)” method. |
| autoAnswer | boolean | Turn on/off auto answer mode.
true : ON false: OFF |
| ctiAutoAnswer | boolean | true : In the cases that an INVITE message has “answer-after=0” at the call-info header, the WebRTC Client automatically answers the call even if its auto answer mode is off(autoAnswer is false). |
| masterVolume | number | Volume of master michrophone.
1000: normal 0: mute Large values over 1001 can be set, but in some cases, depend on environments, clipping noise may happen. |
| language | string | Language preference.If set blank, appropriate language can be defined based on the browser setting.
ja : Japanese is selected. en : English is selected. In the case that the language that is not implemented in WebRTC widget is set as property, English is automatically selected. |
| videoEnabled | boolean | true: Enable Video UI.
false: Disable video UI. Also, as another way, this video UI property can be configured by “Widget.enableWithVideo(boolean)” method. |
| shareStream | boolean | true: In the video call with multiple users, the video stream sent to the first recipient will be shared with subsequent other users. |
| screenCapture | boolean | true: Use screen capturing instead of using camera device. (Only the browsers which support “Screen Capture API“.) |
| bellAudioTarget | string | Specify bell audio output device.If “_all_devices” is set, bell audios will be output by all devices.
If set the deviceId got by MediaDevices.enumerateDevices(), bell audios will be output by the device specified by deviceID. |
| talkAudioTarget | string | Specify voice audio output device.
“_all_devices” is set, voice audios will be output by all devices.(Currently 2019, some browsers haven’t not supported the value “_all_devices“.) If set the deviceId got by “MediaDevices.enumerateDevices()”, voice audios will be output by the device specified by deviceID. |
| configuration | object | Set default value of each key. These values will be used in the case that key values are not set into the argument “configuration” of the function “Phone.startWebRTC()” . |
| callOptions | object | The values will be used as parameters at the “option” argument of the fuction “JsSIP.UA.call” and “JsSIP.RTCSession.answer”<Examples>
Example 1: Enable TURN(TCP) Server {
"callOptions": {
"pcConfig": {
"iceServers": [
{
"urls": "turn:<Domain of TURN Derver>:<Port of TURN Server>?transport=tcp",
"username": "<User name of TURN Server>",
"credential": "<Password of TURN Server>"
}
]
}
}
}
Example 2 Change frame rate of a video {
"callOptions": {
"mediaConstraints": {
"video": {
"frameRate": 2
}
},
"position": {
"videoOptions": {
"call": true,
"answer": true
}
}
}
}
|
| autoplayAllowed | For debug. | |
| autoFocusWindow | For debug. | |
| doNotDisturb | For debug. | |
| multiSession | For debug. | |
| analyserMode | For debug. | |
| getUserMediaTimeout | For debug. | |
| iceCandidateGatheringTimeout | For debug. | |
| defaultOptions | For debug. |