@voicenter-team/opensips-js 1.0.141 → 1.0.142

package/README.md

@@ -205,7 +205,7 @@ Also, there are next public fields on OpensipsJS instance:
 ## Audio
 
 ### Audio methods
-- `initCall(target: String, addToCurrentRoom: Boolean): void` - call to the target. If addToCurrentRoom is true then the call will be added to the user's current room
+- `initCall(target: String, addToCurrentRoom: Boolean, holdOtherCalls: Boolean): void` - call to the target. If addToCurrentRoom is true then the call will be added to the user's current room
 - `holdCall(callId: String, automatic?: Boolean): Promise<void>` - put call on hold
 - `unholdCall(callId: String): Promise<void>` - unhold a call
 - `terminateCall(callId: String): void` - terminate call

package/dist/index.d.ts

@@ -735,6 +735,8 @@ declare interface MSRPSessionExtended extends MSRPSession_2 {
 
 declare type MSRPSessionListener = IncomingMSRPSessionListener | OutgoingMSRPSessionListener;
 
+declare type OnTransportCallback = (parsed: object, message: string) => void
+
 declare interface OpenSIPSEventMap extends UAEventMap {
     ready: readyListener
     connection: connectionListener
@@ -912,6 +914,7 @@ declare interface TriggerListenerOptions {
 
 declare type UAConfigurationExtended = UAConfiguration & {
     overrideUserAgent?: (userAgent: string) => string
+    onTransportCallback?: OnTransportCallback
 }
 
 declare const UAConstructor: typeof UA;
@@ -927,6 +930,7 @@ declare class UAExtended extends UAConstructor implements UAExtendedInterface {
     _janus_sessions: any[];
     protected newStreamPlugins: Array<BaseNewStreamPlugin>;
     protected processStreamPlugins: Array<BaseProcessStreamPlugin>;
+    protected onTransportCallback: OnTransportCallback;
     constructor(configuration: UAConfiguration);
     call(target: string, options?: CallOptionsExtended): RTCSession;
     joinVideoCall(target: string, displayName: string, options: VideoConferenceJoinOptions): any;