## [1.1.56] version 74 - 2020-08-21

### Changed
- 去掉公用stun
- 增加主动挂断判断

build.gradle

@@ -47,12 +47,12 @@ buildscript {
     /**
      * APP版本码
      */
-    ext.app_version_code = 73
+    ext.app_version_code = 74
 
     /**
      * APP版本号
      */
-    ext.app_version = "1.1.55"
+    ext.app_version = "1.1.56"
 
     /**
      * 项目依赖库

home/src/main/code/com/wdkl/ncs/android/component/home/activity/WebRTCVoipAudioActivity.kt

@@ -61,6 +61,7 @@ class WebRTCVoipAudioActivity : Activity(), View.OnClickListener, CallSessionCal
     lateinit var countDownTimer: CountDownTimer
     private var isClick = true //是否可点击
     private var createEnabled = true
+    private var selftHandOff = false
 
     override fun onCreate(savedInstanceState: Bundle?) {
         super.onCreate(savedInstanceState)
@@ -166,7 +167,7 @@ class WebRTCVoipAudioActivity : Activity(), View.OnClickListener, CallSessionCal
                         //6秒后仍然未连接，则断开
                         Handler().postDelayed({
                             Log.i(TAG,"session state:"+gEngineKit?.currentSession?.state+"，userState:"+ SocketManager.getInstance().userState)
-                            if (gEngineKit?.currentSession?.state != EnumType.CallState.Connected){
+                            if (gEngineKit?.currentSession?.state != EnumType.CallState.Connected && !selftHandOff){
                                 showMessage("连接失败，请重试")
 
                                 gEngineKit?.endCall()
@@ -262,6 +263,7 @@ class WebRTCVoipAudioActivity : Activity(), View.OnClickListener, CallSessionCal
                 TcpClient.getInstance().sendMsg(voiceUtilTcpModel.toJson())
             }
         }
+        selftHandOff = true
         gEngineKit?.endCall()
         finish()
     }

readme.md

@@ -26,6 +26,13 @@
 
 ---
 
+## [1.1.56] version 74 - 2020-08-21
+### Changed
+- 去掉公用stun
+- 增加主动挂断判断
+
+---
+
 ## [1.1.55] version 73 - 2020-08-21
 ### Changed
 - 对方挂断逻辑完善

rtc-chat/src/main/java/com/wdkl/skywebrtc/engine/webrtc/WebRTCEngine.java

@@ -514,11 +514,11 @@ public class WebRTCEngine implements IEngine, Peer.IPeerEvent {
     // -----------------------------其他方法--------------------------------
     private void initIceServer() {
         // 初始化一些stun和turn的地址
-        iceServers.add(PeerConnection.IceServer.builder("stun:stun.xten.com:3478").createIceServer());
-        iceServers.add(PeerConnection.IceServer.builder("stun:stun.voipbuster.com:3478").createIceServer());
-        iceServers.add(PeerConnection.IceServer.builder("stun:stun.sipgate.net:3478").createIceServer());
-        iceServers.add(PeerConnection.IceServer.builder("stun:stun.ideasip.com:3478").createIceServer());
-        iceServers.add(PeerConnection.IceServer.builder("stun:stun.internetcalls.com:3478").createIceServer());
+//        iceServers.add(PeerConnection.IceServer.builder("stun:stun.xten.com:3478").createIceServer());
+//        iceServers.add(PeerConnection.IceServer.builder("stun:stun.voipbuster.com:3478").createIceServer());
+//        iceServers.add(PeerConnection.IceServer.builder("stun:stun.sipgate.net:3478").createIceServer());
+//        iceServers.add(PeerConnection.IceServer.builder("stun:stun.ideasip.com:3478").createIceServer());
+//        iceServers.add(PeerConnection.IceServer.builder("stun:stun.internetcalls.com:3478").createIceServer());
         /*
 stun.xten.com
 stun.voipbuster.com