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The Webphone is a SIP over WebRTC Softphone in the browser.

The Webphone component is available as an instant component with no programming required, or can be incrementally hidden and allows entire functionality delivered by the API.

It allows you to create a fully customised solution.

Use Case

Add Audio/Video Bi-directional communication for use in: OTT Mobile Application

  1. Call Center Agent Dashboard
  2. Click-2-Call
  3. Live Website Help
  4. Embedded Applications
  5. Conference Applications




Name Description
onAnswer (detail) Triggers when a call is answered by the other party
onRingingOut (detail) Triggers when 180/183 is receiver from the other party
onRingingIn (detail) Triggers when an incoming call starts ringing
onEnd (cdrDetails) After the call ends, provides the client CDR details
onRegistered () Triggers when SIP registration is successful
onUnregistered () Triggers when SIP is un-registration
onError (error) Triggers on any internal error
this.hide Hides the field Shows the field


Name Description
setConfig ({wsServers: String, username: String, password: String, displayName: String, cli: String, realm: String}) This method is used to set the configuration for the communication component. The following parameters are required:
wsServers: A list of web sockets servers to connect to
username: The username to use to authenticate with the communication server
password: The password to use to authenticate with the communication server
displayName: The display name to use when communicating with other users
cli: The CLI to use for communication
realm: The realm to use for communication
async register() This method is used to register with the communication server. This must be called before making any other calls to the communication component
async unregister() This method is used to un-register from the communication server. This should be called when the communication component is no longer being used
dialpadPress(digit String) This method is used to press a digit on the dial-pad. This can be used to dial a phone number or to enter DTMF tones
answer() This method is used to answer an incoming call
hangup() This method is used to hang up a call
makeCall(address) This method is used to make a call to the specified address
hold() This method is used to put the current call on hold
mute() This method is used to mute the microphone


Name Description Icon
ID This is a unique identifier which is used to access the field by the API and the key of the field when the form is saved
Width [Optional] To set the width of the field
Show (Dialpad)
Show a numeric dial-pad 0-9 # and *
Show (Dialled Number) A text input box containing the number to be dialled
Show (Answer) Show an Answer button
Show (Hang Up) Show a Hang Up button
Show (Hold) Show a Hold button (Places the call on a Network Level Hold)
Show (Mute) Show a Mute Button (Mute microphone channel)
Show (Dial) Show a Dial Button (Which is to be clicked on the start the call)
Show (Status) Show the Registration and Call Status
Auto Register Automatically register when credentials are presented
Default SIP Server The WebRTC socket to connect to (format: wss://, supports failover separated by a comma ','
Default Realm The Authorization Domain (usually the same as the host without the protocol, e.g
Default Display Name The From Name on outgoing calls
Default CLI The From Number on outbound calls
Default Username Username for SIP Authentication
Default Password Password for SIP Authentication
Attribute Action Enable Hidden action to hide the field

First time User?

If you are using the Page Builder components on the ConnexCS platform for the first time, we request you to use our guide on steps to use the Components.