Brekeke Contact Center Suite Wiki

WebRTC Client Options (JSON)

Options for WebRTC Client


Key Type Explanation
ringTone boolean



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



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



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 “” 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.
Yes No
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