支持通用通道对讲，支持全双工对讲

src/main/java/com/genersoft/iot/vmp/gb28181/service/impl/DeviceServiceImpl.java

@@ -574,9 +574,9 @@ public class DeviceServiceImpl implements IDeviceService {
 
     @Override
     public boolean removeCatalogSubscribe(@NotNull Device device, CommonCallback<Boolean> callback) {
-        log.info("[移除目录订阅]: {}", device.getDeviceId());
         String key = SubscribeTaskForCatalog.getKey(device);
         if (subscribeTaskRunner.containsKey(key)) {
+            log.info("[移除目录订阅]: {}", device.getDeviceId());
             SipTransactionInfo transactionInfo = subscribeTaskRunner.getTransactionInfo(key);
             if (transactionInfo == null) {
                 log.warn("[移除目录订阅] 未找到事务信息，{}", device.getDeviceId());
@@ -638,9 +638,9 @@ public class DeviceServiceImpl implements IDeviceService {
 
     @Override
     public boolean removeMobilePositionSubscribe(Device device, CommonCallback<Boolean> callback) {
-        log.info("[移除移动位置订阅]: {}", device.getDeviceId());
         String key = SubscribeTaskForMobilPosition.getKey(device);
         if (subscribeTaskRunner.containsKey(key)) {
+            log.info("[移除移动位置订阅]: {}", device.getDeviceId());
             SipTransactionInfo transactionInfo = subscribeTaskRunner.getTransactionInfo(key);
             if (transactionInfo == null) {
                 log.warn("[移除移动位置订阅] 未找到事务信息，{}", device.getDeviceId());
@@ -703,9 +703,9 @@ public class DeviceServiceImpl implements IDeviceService {
 
     @Override
     public boolean removeAlarmSubscribe(Device device, CommonCallback<Boolean> callback) {
-        log.info("[移除报警订阅]: {}", device.getDeviceId());
         String key = SubscribeTaskForAlarm.getKey(device);
         if (subscribeTaskRunner.containsKey(key)) {
+            log.info("[移除报警订阅]: {}", device.getDeviceId());
             SipTransactionInfo transactionInfo = subscribeTaskRunner.getTransactionInfo(key);
             if (transactionInfo == null) {
                 log.warn("[移除报警订阅] 未找到事务信息，{}", device.getDeviceId());

src/main/java/com/genersoft/iot/vmp/gb28181/service/impl/PlayServiceImpl.java

@@ -1229,6 +1229,12 @@ public class PlayServiceImpl implements IPlayService {
         audioBroadcastResult.setApp(app);
         audioBroadcastResult.setStream(stream);
         audioBroadcastResult.setStreamInfo(new StreamContent(mediaServerService.getStreamInfoByAppAndStream(mediaServerItem, app, stream, null, null, null, false)));
+        if (!broadcastMode) {
+            audioBroadcastResult.setPlayStreamInfo(new StreamContent(
+                    mediaServerService.getStreamInfoByAppAndStream(mediaServerItem,
+                            MediaStreamUtil.GB28181_TALK, stream + "_talk",
+                            null, null, null, true)));
+        }
         audioBroadcastResult.setCodec("G.711");
         return audioBroadcastResult;
     }

src/main/java/com/genersoft/iot/vmp/gb28181/service/impl/SourceBroadcastServiceForGbImpl.java

@@ -1,8 +1,6 @@
 package com.genersoft.iot.vmp.gb28181.service.impl;
 
 import com.genersoft.iot.vmp.common.enums.ChannelDataType;
-import com.genersoft.iot.vmp.common.enums.MediaStreamUtil;
-import com.genersoft.iot.vmp.conf.UserSetting;
 import com.genersoft.iot.vmp.conf.exception.ControllerException;
 import com.genersoft.iot.vmp.gb28181.bean.CommonGBChannel;
 import com.genersoft.iot.vmp.gb28181.bean.Device;
@@ -11,12 +9,9 @@ import com.genersoft.iot.vmp.gb28181.service.IDeviceChannelService;
 import com.genersoft.iot.vmp.gb28181.service.IDeviceService;
 import com.genersoft.iot.vmp.gb28181.service.IPlayService;
 import com.genersoft.iot.vmp.gb28181.service.ISourceBroadcastService;
-import com.genersoft.iot.vmp.media.bean.MediaServer;
-import com.genersoft.iot.vmp.media.service.IMediaServerService;
 import com.genersoft.iot.vmp.vmanager.bean.AudioBroadcastResult;
 import com.genersoft.iot.vmp.vmanager.bean.AudioTalkResult;
 import com.genersoft.iot.vmp.vmanager.bean.ErrorCode;
-import com.genersoft.iot.vmp.vmanager.bean.StreamContent;
 import lombok.extern.slf4j.Slf4j;
 import org.springframework.beans.factory.annotation.Autowired;
 import org.springframework.stereotype.Service;
@@ -34,12 +29,6 @@ public class SourceBroadcastServiceForGbImpl implements ISourceBroadcastService
     @Autowired
     private IDeviceChannelService deviceChannelService;
 
-    @Autowired
-    private IMediaServerService mediaServerService;
-
-    @Autowired
-    private UserSetting userSetting;
-
     @Override
     public AudioTalkResult startBroadcast(CommonGBChannel channel) {
         Device device = deviceService.getDevice(channel.getDataDeviceId());
@@ -79,14 +68,9 @@ public class SourceBroadcastServiceForGbImpl implements ISourceBroadcastService
         }
         AudioBroadcastResult abResult = playService.audioBroadcast(
                 device.getDeviceId(), deviceChannel.getDeviceId(), false);
-        MediaServer mediaServer = mediaServerService.getMediaServerForMinimumLoad(null);
-        StreamContent playStream = new StreamContent(
-                mediaServerService.getStreamInfoByAppAndStream(mediaServer,
-                        MediaStreamUtil.GB28181_TALK, abResult.getStream() + "_talk",
-                        null, null, null, false));
         AudioTalkResult result = new AudioTalkResult();
         result.setPushStream(abResult.getStreamInfo());
-        result.setPlayStream(playStream);
+        result.setPlayStream(abResult.getPlayStreamInfo());
         return result;
     }
 

src/main/java/com/genersoft/iot/vmp/gb28181/transmit/cmd/impl/SIPCommander.java

@@ -496,7 +496,7 @@ public class SIPCommander implements ISIPCommander {
         }
         if (!mediaServerItem.isRtpEnable()) {
             // 单端口暂不支持语音喊话
-            log.info("[语音喊话] 单端口暂不支持此操作");
+            log.warn("[语音喊话] 单端口暂不支持此操作");
             return;
         }
 

src/main/java/com/genersoft/iot/vmp/vmanager/bean/AudioBroadcastResult.java

@@ -1,8 +1,13 @@
 package com.genersoft.iot.vmp.vmanager.bean;
 
+import lombok.Getter;
+import lombok.Setter;
+
 /**
  * @author lin
  */
+@Setter
+@Getter
 public class AudioBroadcastResult {
     /**
      * 推流的各个方式流地址
@@ -24,36 +29,10 @@ public class AudioBroadcastResult {
      */
     private String stream;
 
+    /**
+     * 播放流地址（设备音频通过ZLM播放给浏览器），对讲时设置
+     */
+    private StreamContent playStreamInfo;
 
-    public StreamContent getStreamInfo() {
-        return streamInfo;
-    }
 
-    public void setStreamInfo(StreamContent streamInfo) {
-        this.streamInfo = streamInfo;
-    }
-
-    public String getCodec() {
-        return codec;
-    }
-
-    public void setCodec(String codec) {
-        this.codec = codec;
-    }
-
-    public String getApp() {
-        return app;
-    }
-
-    public void setApp(String app) {
-        this.app = app;
-    }
-
-    public String getStream() {
-        return stream;
-    }
-
-    public void setStream(String stream) {
-        this.stream = stream;
-    }
 }

web/src/views/channel/audioTalk.vue

@@ -355,7 +355,9 @@ export default {
 
       this.talkAudioRtc.on(ZLMRTCClient.Events.WEBRTC_ON_CONNECTION_STATE_CHANGE, (s) => {
         console.warn('[ChAudioTalk] 音频播放连接状态:', s)
-        if (s === 'disconnected' || s === 'failed' || s === 'closed') {
+        if (s === 'connected') {
+          this.playConnected = true
+        } else if (s === 'disconnected' || s === 'failed' || s === 'closed') {
           this.playConnected = false
           this.talkAudioFailed = true
           if (this.talkStatus === 1) {

web/src/views/device/dialog/audioTalk.vue

@@ -51,9 +51,19 @@
             />
             <p style="margin-top: 16px; color: #606266;">
               <span v-if="talkStatus === -2">正在释放资源</span>
-              <span v-if="talkStatus === -1">点击开始{{ talkMode ? '对讲' : '喊话' }}</span>
+              <span v-if="talkStatus === -1">点击开始{{ talkMode ? '喊话' : '对讲' }}</span>
               <span v-if="talkStatus === 0">等待接通中...</span>
-              <span v-if="talkStatus === 1">请说话</span>
+              <span v-if="talkStatus === 1 && talkMode">喊话中</span>
+              <span v-if="talkStatus === 1 && !talkMode && !playConnected">等待接通中...</span>
+              <span v-if="talkStatus === 1 && !talkMode && playConnected">对讲中</span>
+            </p>
+            <p v-if="talkStatus === 1 && !talkMode && talkAudioFailed" style="margin-top: 8px;">
+              <el-button
+                type="warning"
+                size="mini"
+                icon="el-icon-refresh"
+                @click="retryTalkAudio"
+              >重试音频</el-button>
             </p>
           </div>
         </div>
@@ -129,7 +139,10 @@ export default {
       if (this.talkStatus === -2) return 'primary'
       if (this.talkStatus === -1) return 'primary'
       if (this.talkStatus === 0) return 'warning'
-      if (this.talkStatus === 1) return 'danger'
+      if (this.talkStatus === 1) {
+        if (!this.talkMode && !this.playConnected) return 'warning'
+        return 'danger'
+      }
     },
     async talkButtonClick() {
       if (this.talkStatus === -1) {
@@ -145,6 +158,13 @@ export default {
         const si = data.streamInfo
         const url = document.location.protocol.includes('https') ? si.rtcs : si.rtc
         this.startWebrtcPush(url)
+
+        const playStreamInfo = data?.playStreamInfo
+        if (!this.talkMode && playStreamInfo) {
+          this.talkAudioPlayStream = playStreamInfo
+          this.startTalkAudioPlay(playStreamInfo)
+          this.muteVideoPlayer()
+        }
       } catch (e) {
         this.$message({ showClose: true, message: e, type: 'error' })
         this.talkStatus = -1
@@ -181,6 +201,105 @@ export default {
         })
         .catch(() => { this.talkStatus = -1 })
     },
+    muteVideoPlayer() {
+      const player = this.$refs.playerTabs
+      if (!player) return
+      if (player.mute) {
+        player.mute()
+      }
+    },
+    unmuteVideoPlayer() {
+      const player = this.$refs.playerTabs
+      if (!player) return
+      if (player.cancelMute) {
+        player.cancelMute()
+      }
+    },
+    startTalkAudioPlay(playStreamInfo) {
+      if (this.talkAudioRtc) {
+        this.talkAudioRtc.close()
+      }
+      if (this.talkAudioRetryTimer) {
+        clearTimeout(this.talkAudioRetryTimer)
+      }
+
+      const url = location.protocol === 'https:' ? playStreamInfo.rtcs : playStreamInfo.rtc
+      if (!url) {
+        console.warn('[AudioTalk] 无可用的设备音频播放地址')
+        return
+      }
+      this.talkAudioRetryTimer = setTimeout(() => {
+        this.pollMediaInfoAndPlay(playStreamInfo)
+      }, 800)
+    },
+    async pollMediaInfoAndPlay(playStreamInfo) {
+      try {
+        const data = await this.$store.dispatch('server/getMediaInfo', {
+          app: playStreamInfo.app,
+          stream: playStreamInfo.stream,
+          mediaServerId: playStreamInfo.mediaServerId
+        })
+        if (data) {
+          const url = location.protocol === 'https:' ? playStreamInfo.rtcs : playStreamInfo.rtc
+          this.startTalkAudioByRtc(url)
+        } else {
+          throw new Error('no data')
+        }
+      } catch (e) {
+        if (this.talkStatus === 1 || this.talkStatus === 0) {
+          this.talkAudioRetryTimer = setTimeout(() => {
+            this.pollMediaInfoAndPlay(playStreamInfo)
+          }, 800)
+        }
+      }
+    },
+    startTalkAudioByRtc(url) {
+      this.talkAudioFailed = false
+      this.talkAudioRtc = new ZLMRTCClient.Endpoint({
+        debug: false,
+        element: document.getElementById('audioTalkVideo'),
+        zlmsdpUrl: url,
+        simulecast: false,
+        useCamera: false,
+        audioEnable: true,
+        videoEnable: false,
+        recvOnly: true,
+        usedatachannel: false
+      })
+
+      this.talkAudioRtc.on(ZLMRTCClient.Events.WEBRTC_OFFER_ANWSER_EXCHANGE_FAILED, (e) => {
+        console.warn('[AudioTalk] 播放流offer失败:', e?.code, e?.msg)
+        if (e && e.code == -400 && e.msg == '流不存在') {
+          this.talkAudioRetryTimer = setTimeout(() => {
+            this.startTalkAudioByRtc(url)
+          }, 1000)
+        }
+      })
+
+      this.talkAudioRtc.on(ZLMRTCClient.Events.WEBRTC_ON_REMOTE_STREAMS, () => {
+        console.warn('[AudioTalk] 设备音频流到达')
+        this.playConnected = true
+      })
+
+      this.talkAudioRtc.on(ZLMRTCClient.Events.WEBRTC_ICE_CANDIDATE_ERROR, () => {
+        console.error('[AudioTalk] 音频播放ICE协商失败')
+      })
+
+      this.talkAudioRtc.on(ZLMRTCClient.Events.WEBRTC_ON_CONNECTION_STATE_CHANGE, (s) => {
+        console.warn('[AudioTalk] 音频播放连接状态:', s)
+        if (s === 'connected') {
+          this.playConnected = true
+        } else if (s === 'disconnected' || s === 'failed' || s === 'closed') {
+          this.playConnected = false
+          this.talkAudioFailed = true
+          if (this.talkStatus === 1) {
+            this.talkAudioRetryTimer = setTimeout(() => {
+              this.startTalkAudioByRtc(url)
+            }, 2000)
+          }
+        }
+      })
+    },
     async stopTalk() {
       this.talkStatus = -2
       if (this.broadcastRtc) {
@@ -198,6 +317,7 @@ export default {
       this.talkAudioFailed = false
       this.talkAudioPlayStream = null
       this.playConnected = false
+      this.unmuteVideoPlayer()
       try {
         await this.$store.dispatch('play/broadcastStop', [this.deviceId, this.channelId])
       } catch (e) {
@@ -205,6 +325,11 @@ export default {
       }
       this.talkStatus = -1
     },
+    retryTalkAudio() {
+      if (this.talkAudioPlayStream) {
+        this.startTalkAudioPlay(this.talkAudioPlayStream)
+      }
+    },
     close() {
       if (this.showPlayer && this.$refs.playerTabs) {
         this.$refs.playerTabs.stop()