IncomingCallInterface

Returns the callee address information related the call instance

Returns caller name of the related call

Returns the caller address information related the call instance

This method sets the caller name of origin. Especially needed for anonymous calls.

Returns callId that related with SIP session id For the outgoing call, this value is not set until establish call is succeeded

Returns the state of the call

Sets given custom SIP Headers as custom parameters to call in order to send them in all mid-call event requests.

Gets custom SIP headers as a key-value map gathered from call notification. Currently supported custom SIP headers : <li>x-nt-service</li>

Returns unique identifier of call object which is generated from SDK Identifier is NOT related with SIP session id

Returns if call is in mute state

Sets video enabled or not for this call.

Sets microphone muted or not for this call.

Returns the media attributes of the call

Returns type of the Video View which is on front.

Accepts the incoming call

Accepts the incoming call and sends the given custom SIP headers to WebRTC GW server.

In case of callerAddress or calleeAddress changes this callback will notify the application.

Puts Video View of given type in front of any other Video View.

Indicates remote side can send video or not. Local side can receive video if he wishes. It is determined by Remote Session Description direction as follows

Indicates remote side can receive video or not. Local side can send video if he wishes. It is determined by Remote Session Description direction as follows

Changes video resolution as given values. These values are not saved so they can be thought as temporary

Transfers the current call to the transfer target given with another call session in a consultative transfer manner. ** Note that, consultative transfer operation cannot be used with Anonymous calls.**

Stops the related call and releases allocated resources.

Stops the related call with reason and releases allocated resources. If reason string char length is more than 600 char, first 600 char will send to remote.

Forwards the related incoming call to another terminator

Returns the list of available and supported codecs

Gets RTP/RTP Statistics for related call

Holds the related call.

Ignores the related incoming call

Mutes the voice of local user.

Rejects the related incoming call

Sends the given custom SIP headers to WebRTC GW server.

Inserts Dual Tone Multi Frequency Signal to sending queue and returns the success

Sends the ringing feedback to WebRTC GW. Needs to be called if Ringing feedback enabled.

Allows applications to set AudioSourceType.

Sets device capture to image source for the related call

Sets device capture and video resolution to specified position and resolution of the related call

Allows applications to set the audio file as external audio source and play it once. SDK can transmit audio file to remote peer if application uses this API.

Allows applications to set the audio file as external audio source and play it for the specified times. SDK can transmit audio file to remote peer if application uses this API.

Sets GUI View for video of local user

Sets GUI View for video of remote peer

Takes still image of the active local video stream

Transfers the call to the given address. ** Note that, transfer operation cannot be used with Anonymous calls.**

Unholds the related call.

Opens the voice of local user.

Starts video of local user.

Stops the video of local user.