Merge branch 'talk' into main-dev

# Conflicts:
#	src/main/java/com/genersoft/iot/vmp/conf/redis/RedisConfig.java
#	src/main/java/com/genersoft/iot/vmp/gb28181/bean/AudioBroadcastCatch.java
#	src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/InviteRequestProcessor.java
#	src/main/java/com/genersoft/iot/vmp/media/zlm/ZLMHttpHookListener.java
#	src/main/java/com/genersoft/iot/vmp/service/IPlayService.java
#	src/main/java/com/genersoft/iot/vmp/service/impl/PlayServiceImpl.java

src/main/java/com/genersoft/iot/vmp/gb28181/bean/AudioBroadcastCatch.java

@@ -63,6 +63,8 @@ public class AudioBroadcastCatch {
      */
     private SipTransactionInfo sipTransactionInfo;
 
+    private MediaServerItem mediaServerItem;
+
 
     public String getDeviceId() {
         return deviceId;
@@ -123,4 +125,12 @@ public class AudioBroadcastCatch {
     public void setSipTransactionInfoByRequset(SIPResponse response) {
         this.sipTransactionInfo = new SipTransactionInfo(response, false);
     }
+
+    public MediaServerItem getMediaServerItem() {
+        return mediaServerItem;
+    }
+
+    public void setMediaServerItem(MediaServerItem mediaServerItem) {
+        this.mediaServerItem = mediaServerItem;
+    }
 }

src/main/java/com/genersoft/iot/vmp/gb28181/bean/SendRtpItem.java

@@ -49,7 +49,7 @@ public class SendRtpItem {
     /**
      * 设备推流的streamId
      */
-    private String streamId;
+    private String stream;
 
     /**
      * 是否为tcp
@@ -117,6 +117,11 @@ public class SendRtpItem {
      */
     private InviteStreamType playType;
 
+    /**
+     * 发流的同时收流
+     */
+    private String receiveStream;
+
     public String getIp() {
         return ip;
     }
@@ -181,12 +186,12 @@ public class SendRtpItem {
         this.app = app;
     }
 
-    public String getStreamId() {
+    public String getStream() {
-        return streamId;
+        return stream;
     }
 
-    public void setStreamId(String streamId) {
+    public void setStream(String stream) {
-        this.streamId = streamId;
+        this.stream = stream;
     }
 
     public boolean isTcp() {
@@ -292,4 +297,12 @@ public class SendRtpItem {
     public void setRtcp(boolean rtcp) {
         this.rtcp = rtcp;
     }
+
+    public String getReceiveStream() {
+        return receiveStream;
+    }
+
+    public void setReceiveStream(String receiveStream) {
+        this.receiveStream = receiveStream;
+    }
 }

src/main/java/com/genersoft/iot/vmp/gb28181/event/SipSubscribe.java

@@ -1,6 +1,5 @@
 package com.genersoft.iot.vmp.gb28181.event;
 
-import com.genersoft.iot.vmp.conf.exception.SsrcTransactionNotFoundException;
 import com.genersoft.iot.vmp.gb28181.bean.DeviceNotFoundEvent;
 import gov.nist.javax.sip.message.SIPRequest;
 import org.slf4j.Logger;

src/main/java/com/genersoft/iot/vmp/gb28181/session/VideoStreamSessionManager.java

@@ -29,7 +29,8 @@ public class VideoStreamSessionManager {
 		play,
 		playback,
 		download,
-		broadcast
+		broadcast,
+		talk
 	}
 
 	/**

src/main/java/com/genersoft/iot/vmp/gb28181/task/SipRunner.java

@@ -94,12 +94,12 @@ public class SipRunner implements CommandLineRunner {
         if (sendRtpItems.size() > 0) {
             for (SendRtpItem sendRtpItem : sendRtpItems) {
                 MediaServerItem mediaServerItem = mediaServerService.getOne(sendRtpItem.getMediaServerId());
-                redisCatchStorage.deleteSendRTPServer(sendRtpItem.getPlatformId(),sendRtpItem.getChannelId(), sendRtpItem.getCallId(),sendRtpItem.getStreamId());
+                redisCatchStorage.deleteSendRTPServer(sendRtpItem.getPlatformId(),sendRtpItem.getChannelId(), sendRtpItem.getCallId(),sendRtpItem.getStream());
                 if (mediaServerItem != null) {
                     Map<String, Object> param = new HashMap<>();
                     param.put("vhost","__defaultVhost__");
                     param.put("app",sendRtpItem.getApp());
-                    param.put("stream",sendRtpItem.getStreamId());
+                    param.put("stream",sendRtpItem.getStream());
                     param.put("ssrc",sendRtpItem.getSsrc());
                     JSONObject jsonObject = zlmresTfulUtils.stopSendRtp(mediaServerItem, param);
                     if (jsonObject != null && jsonObject.getInteger("code") == 0) {

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

@@ -2,10 +2,7 @@ package com.genersoft.iot.vmp.gb28181.transmit.cmd;
 
 import com.genersoft.iot.vmp.common.StreamInfo;
 import com.genersoft.iot.vmp.conf.exception.SsrcTransactionNotFoundException;
-import com.genersoft.iot.vmp.gb28181.bean.Device;
+import com.genersoft.iot.vmp.gb28181.bean.*;
-import com.genersoft.iot.vmp.gb28181.bean.DeviceAlarm;
-import com.genersoft.iot.vmp.gb28181.bean.InviteStreamCallback;
-import com.genersoft.iot.vmp.gb28181.bean.SipTransactionInfo;
 import com.genersoft.iot.vmp.gb28181.event.SipSubscribe;
 import com.genersoft.iot.vmp.media.zlm.ZlmHttpHookSubscribe;
 import com.genersoft.iot.vmp.media.zlm.dto.MediaServerItem;
@@ -131,7 +128,7 @@ public interface ISIPCommander {
 	 */
 	void streamByeCmd(Device device, String channelId, String stream, String callId, SipSubscribe.Event okEvent) throws InvalidArgumentException, SipException, ParseException, SsrcTransactionNotFoundException;
 
-	void talkStreamCmd(MediaServerItem mediaServerItem, SSRCInfo ssrcInfo, Device device, String channelId, String callId, ZlmHttpHookSubscribe.Event event, ZlmHttpHookSubscribe.Event eventForPush, SipSubscribe.Event okEvent, SipSubscribe.Event errorEvent) throws InvalidArgumentException, SipException, ParseException;
+	void talkStreamCmd(MediaServerItem mediaServerItem, SendRtpItem sendRtpItem, Device device, String channelId, String callId, ZlmHttpHookSubscribe.Event event, ZlmHttpHookSubscribe.Event eventForPush, SipSubscribe.Event okEvent, SipSubscribe.Event errorEvent) throws InvalidArgumentException, SipException, ParseException;
 
 
 	void streamByeCmd(Device device, String channelId, String stream, String callId) throws InvalidArgumentException, ParseException, SipException, SsrcTransactionNotFoundException;

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

@@ -32,7 +32,6 @@ import org.springframework.beans.factory.annotation.Autowired;
 import org.springframework.context.annotation.DependsOn;
 import org.springframework.stereotype.Component;
 import org.springframework.util.ObjectUtils;
-import org.springframework.util.StringUtils;
 
 import javax.sip.InvalidArgumentException;
 import javax.sip.ResponseEvent;
@@ -584,9 +583,9 @@ public class SIPCommander implements ISIPCommander {
     }
 
     @Override
-    public void talkStreamCmd(MediaServerItem mediaServerItem, SSRCInfo ssrcInfo, Device device, String channelId, String callId, ZlmHttpHookSubscribe.Event event, ZlmHttpHookSubscribe.Event eventForPush, SipSubscribe.Event okEvent, SipSubscribe.Event errorEvent) throws InvalidArgumentException, SipException, ParseException {
+    public void talkStreamCmd(MediaServerItem mediaServerItem, SendRtpItem sendRtpItem, Device device, String channelId, String callId, ZlmHttpHookSubscribe.Event event, ZlmHttpHookSubscribe.Event eventForPush, SipSubscribe.Event okEvent, SipSubscribe.Event errorEvent) throws InvalidArgumentException, SipException, ParseException {
 
-        String stream = ssrcInfo.getStream();
+        String stream = sendRtpItem.getStream();
 
         if (device == null) {
             return;
@@ -597,7 +596,7 @@ public class SIPCommander implements ISIPCommander {
             return;
         }
 
-        logger.info("[语音对讲] {} 分配的ZLM为: {} [{}:{}]", stream, mediaServerItem.getId(), mediaServerItem.getIp(), ssrcInfo.getPort());
+        logger.info("[语音对讲] {} 分配的ZLM为: {} [{}:{}]", stream, mediaServerItem.getId(), mediaServerItem.getIp(), sendRtpItem.getPort());
         HookSubscribeForStreamChange hookSubscribeForStreamChange = HookSubscribeFactory.on_stream_changed("rtp", stream, true, "rtsp", mediaServerItem.getId());
         subscribe.addSubscribe(hookSubscribeForStreamChange, (MediaServerItem mediaServerItemInUse, JSONObject json) -> {
             if (event != null) {
@@ -622,24 +621,27 @@ public class SIPCommander implements ISIPCommander {
         content.append("c=IN IP4 " + mediaServerItem.getSdpIp() + "\r\n");
         content.append("t=0 0\r\n");
 
-        content.append("m=audio " + ssrcInfo.getPort() + " RTP/AVP 8\r\n");
+        content.append("m=audio " + sendRtpItem.getPort() + " TCP/RTP/AVP 8\r\n");
+        content.append("a=setup:passive\r\n");
+        content.append("a=connection:new\r\n");
         content.append("a=sendrecv\r\n");
         content.append("a=rtpmap:8 PCMA/8000\r\n");
 
-        content.append("y=" + ssrcInfo.getSsrc() + "\r\n");//ssrc
+        content.append("y=" + sendRtpItem.getSsrc() + "\r\n");//ssrc
         // f字段:f= v/编码格式/分辨率/帧率/码率类型/码率大小a/编码格式/码率大小/采样率
         content.append("f=v/////a/1/8/1" + "\r\n");
 
-        Request request = headerProvider.createInviteRequest(device, channelId, content.toString(), SipUtils.getNewViaTag(), SipUtils.getNewFromTag(), null, ssrcInfo.getSsrc(), callIdHeader);
+        Request request = headerProvider.createInviteRequest(device, channelId, content.toString(),
+                SipUtils.getNewViaTag(), SipUtils.getNewFromTag(), null, sendRtpItem.getSsrc(), callIdHeader);
         sipSender.transmitRequest(sipLayer.getLocalIp(device.getLocalIp()), request, (e -> {
-            streamSession.remove(device.getDeviceId(), channelId, ssrcInfo.getStream());
+            streamSession.remove(device.getDeviceId(), channelId, sendRtpItem.getStream());
-            mediaServerService.releaseSsrc(mediaServerItem.getId(), ssrcInfo.getSsrc());
+            mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
             errorEvent.response(e);
         }), e -> {
             // 这里为例避免一个通道的点播只有一个callID这个参数使用一个固定值
             ResponseEvent responseEvent = (ResponseEvent) e.event;
             SIPResponse response = (SIPResponse) responseEvent.getResponse();
-            streamSession.put(device.getDeviceId(), channelId, "talk", stream, ssrcInfo.getSsrc(), mediaServerItem.getId(), response, VideoStreamSessionManager.SessionType.play);
+            streamSession.put(device.getDeviceId(), channelId, "talk", stream, sendRtpItem.getSsrc(), mediaServerItem.getId(), response, VideoStreamSessionManager.SessionType.talk);
             okEvent.response(e);
         });
     }

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

@@ -694,7 +694,7 @@ public class SIPCommanderFroPlatform implements ISIPCommanderForPlatform {
         MediaServerItem mediaServerItem = mediaServerService.getOne(mediaServerId);
         if (mediaServerItem != null) {
             mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
-            zlmrtpServerFactory.closeRtpServer(mediaServerItem, sendRtpItem.getStreamId());
+            zlmrtpServerFactory.closeRtpServer(mediaServerItem, sendRtpItem.getStream());
         }
         SIPRequest byeRequest = headerProviderPlatformProvider.createByeRequest(platform, sendRtpItem);
         if (byeRequest == null) {

src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/AckRequestProcessor.java

@@ -102,12 +102,12 @@ public class AckRequestProcessor extends SIPRequestProcessorParent implements In
 			}
 			String isUdp = sendRtpItem.isTcp() ? "0" : "1";
 			MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
-			logger.info("收到ACK，rtp/{}开始向上级推流, 目标={}:{}，SSRC={}, RTCP={}", sendRtpItem.getStreamId(),
+			logger.info("收到ACK，rtp/{}开始向上级推流, 目标={}:{}，SSRC={}, RTCP={}", sendRtpItem.getStream(),
 					sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isRtcp());
 			Map<String, Object> param = new HashMap<>(12);
 			param.put("vhost","__defaultVhost__");
 			param.put("app",sendRtpItem.getApp());
-			param.put("stream",sendRtpItem.getStreamId());
+			param.put("stream",sendRtpItem.getStream());
 			param.put("ssrc", sendRtpItem.getSsrc());
 			param.put("src_port", sendRtpItem.getLocalPort());
 			param.put("pt", sendRtpItem.getPt());
@@ -121,7 +121,7 @@ public class AckRequestProcessor extends SIPRequestProcessorParent implements In
 
 			if (mediaInfo == null) {
 				RequestPushStreamMsg requestPushStreamMsg = RequestPushStreamMsg.getInstance(
-						sendRtpItem.getMediaServerId(), sendRtpItem.getApp(), sendRtpItem.getStreamId(),
+						sendRtpItem.getMediaServerId(), sendRtpItem.getApp(), sendRtpItem.getStream(),
 						sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isTcp(),
 						sendRtpItem.getLocalPort(), sendRtpItem.getPt(), sendRtpItem.isUsePs(), sendRtpItem.isOnlyAudio());
 				redisGbPlayMsgListener.sendMsgForStartSendRtpStream(sendRtpItem.getServerId(), requestPushStreamMsg, json -> {

src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/ByeRequestProcessor.java

@@ -97,7 +97,7 @@ public class ByeRequestProcessor extends SIPRequestProcessorParent implements In
 
 		if (sendRtpItem != null){
 			logger.info("[收到bye] {}/{}", sendRtpItem.getPlatformId(), sendRtpItem.getChannelId());
-			String streamId = sendRtpItem.getStreamId();
+			String streamId = sendRtpItem.getStream();
 			MediaServerItem mediaServerItem = mediaServerService.getOne(sendRtpItem.getMediaServerId());
 			if (mediaServerItem == null) {
 				return;
@@ -105,7 +105,7 @@ public class ByeRequestProcessor extends SIPRequestProcessorParent implements In
 
 			Boolean ready = zlmrtpServerFactory.isStreamReady(mediaServerItem, sendRtpItem.getApp(), streamId);
 			if (!ready) {
-				logger.info("[收到bye] 发现流{}/{}已经结束，不需处理", sendRtpItem.getApp(), sendRtpItem.getStreamId());
+				logger.info("[收到bye] 发现流{}/{}已经结束，不需处理", sendRtpItem.getApp(), sendRtpItem.getStream());
 				return;
 			}
 			Map<String, Object> param = new HashMap<>();
@@ -113,7 +113,7 @@ public class ByeRequestProcessor extends SIPRequestProcessorParent implements In
 			param.put("app",sendRtpItem.getApp());
 			param.put("stream",streamId);
 			param.put("ssrc",sendRtpItem.getSsrc());
-			logger.info("[收到bye] 停止向上级推流：{}", streamId);
+			logger.info("[收到bye] 停止推流：{}", streamId);
 			MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
 			redisCatchStorage.deleteSendRTPServer(sendRtpItem.getPlatformId(), sendRtpItem.getChannelId(), callIdHeader.getCallId(), null);
 			zlmrtpServerFactory.stopSendRtpStream(mediaInfo, param);
@@ -129,15 +129,14 @@ public class ByeRequestProcessor extends SIPRequestProcessorParent implements In
 					try {
 						logger.warn("[停止点播] {}/{}", sendRtpItem.getDeviceId(), sendRtpItem.getChannelId());
 						cmder.streamByeCmd(device, sendRtpItem.getChannelId(), streamId, null);
-					} catch (InvalidArgumentException | ParseException | SipException |
+					} catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
-							 SsrcTransactionNotFoundException e) {
 						logger.error("[收到bye] {} 无其它观看者，通知设备停止推流， 发送BYE失败 {}",streamId, e.getMessage());
 					}
 				}
 
 				if (sendRtpItem.getPlayType().equals(InviteStreamType.PUSH)) {
 					MessageForPushChannel messageForPushChannel = MessageForPushChannel.getInstance(0,
-							sendRtpItem.getApp(), sendRtpItem.getStreamId(), sendRtpItem.getChannelId(),
+							sendRtpItem.getApp(), sendRtpItem.getStream(), sendRtpItem.getChannelId(),
 							sendRtpItem.getPlatformId(), null, null, sendRtpItem.getMediaServerId());
 					redisCatchStorage.sendStreamPushRequestedMsg(messageForPushChannel);
 				}

src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/InviteRequestProcessor.java

@@ -478,7 +478,7 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
                     if ("Playback".equalsIgnoreCase(sessionName)) {
                         sendRtpItem.setPlayType(InviteStreamType.PLAYBACK);
                         SSRCInfo ssrcInfo = mediaServerService.openRTPServer(mediaServerItem, null, device.isSsrcCheck(), true);
-                        sendRtpItem.setStreamId(ssrcInfo.getStream());
+                        sendRtpItem.setStream(ssrcInfo.getStream());
                         // 写入redis， 超时时回复
                         redisCatchStorage.updateSendRTPSever(sendRtpItem);
                         playService.playBack(mediaServerItem, ssrcInfo, device.getDeviceId(), channelId, DateUtil.formatter.format(start),
@@ -523,7 +523,7 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
                             }
                             SSRCInfo ssrcInfo = mediaServerService.openRTPServer(mediaServerItem, streamId, null, device.isSsrcCheck(), false);
                             logger.info(JSONObject.toJSONString(ssrcInfo));
-                            sendRtpItem.setStreamId(ssrcInfo.getStream());
+                            sendRtpItem.setStream(ssrcInfo.getStream());
                             sendRtpItem.setSsrc(ssrc.equals(ssrcDefault) ? ssrcInfo.getSsrc() : ssrc);
 
                             // 写入redis， 超时时回复
@@ -533,12 +533,12 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
                                 redisCatchStorage.deleteSendRTPServer(platform.getServerGBId(), finalChannelId, callIdHeader.getCallId(), null);
                             });
                         } else {
-                            sendRtpItem.setStreamId(playTransaction.getStream());
+                            sendRtpItem.setStream(playTransaction.getStream());
                             // 写入redis， 超时时回复
                             redisCatchStorage.updateSendRTPSever(sendRtpItem);
                             JSONObject jsonObject = new JSONObject();
                             jsonObject.put("app", sendRtpItem.getApp());
-                            jsonObject.put("stream", sendRtpItem.getStreamId());
+                            jsonObject.put("stream", sendRtpItem.getStream());
                             hookEvent.response(mediaServerItem, jsonObject);
                         }
                     }
@@ -982,6 +982,21 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
                     }
                     return;
                 }
+                String addressStr = sdp.getOrigin().getAddress();
+                logger.info("设备{}请求语音流，地址：{}:{}，ssrc：{}, {}", requesterId, addressStr, port, ssrc,
+                        mediaTransmissionTCP ? (tcpActive? "TCP主动":"TCP被动") : "UDP");
+
+                MediaServerItem mediaServerItem = audioBroadcastCatch.getMediaServerItem();
+                if (mediaServerItem == null) {
+                    logger.warn("未找到语音喊话使用的zlm");
+                    try {
+                        responseAck(request, Response.BUSY_HERE);
+                    } catch (SipException | InvalidArgumentException | ParseException e) {
+                        logger.error("[命令发送失败] invite 未找到可用的zlm: {}", e.getMessage());
+                        playService.stopAudioBroadcast(device.getDeviceId(), audioBroadcastCatch.getChannelId());
+                    }
+                    return;
+                }
                 String addressStr = sdp.getConnection().getAddress();
                 logger.info("设备{}请求语音流， 收流地址：{}:{}，ssrc：{}, {}, 对讲方式：{}", requesterId, addressStr, port, ssrc,
                         mediaTransmissionTCP ? (tcpActive? "TCP主动":"TCP被动") : "UDP", sdp.getSessionName().getValue());

src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/info/InfoRequestProcessor.java

@@ -102,7 +102,7 @@ public class InfoRequestProcessor extends SIPRequestProcessorParent implements I
                 String contentSubType = header.getContentSubType();
                 if ("Application".equalsIgnoreCase(contentType) && "MANSRTSP".equalsIgnoreCase(contentSubType)) {
                     SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(null, null, null, callIdHeader.getCallId());
-                    String streamId = sendRtpItem.getStreamId();
+                    String streamId = sendRtpItem.getStream();
                     StreamInfo streamInfo = redisCatchStorage.queryPlayback(null, null, streamId, null);
                     if (null == streamInfo) {
                         responseAck(request, Response.NOT_FOUND, "stream " + streamId + " not found");

src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/message/notify/cmd/MediaStatusNotifyMessageHandler.java

@@ -90,7 +90,7 @@ public class MediaStatusNotifyMessageHandler extends SIPRequestProcessorParent i
 
                 try {
                     cmder.streamByeCmd(device, ssrcTransaction.getChannelId(), null, callIdHeader.getCallId());
-                } catch (InvalidArgumentException | ParseException | SsrcTransactionNotFoundException | SipException e) {
+                } catch (InvalidArgumentException | ParseException  | SipException | SsrcTransactionNotFoundException e) {
                     logger.error("[录像流]推送完毕，收到关流通知， 发送BYE失败 {}", e.getMessage());
                 }
 

src/main/java/com/genersoft/iot/vmp/gb28181/utils/SipUtils.java

@@ -123,7 +123,7 @@ public class SipUtils {
     }
 
     public static String getNewCallId() {
-        return (int) Math.floor(Math.random() * 10000) + "";
+        return (int) Math.floor(Math.random() * 1000000000) + "";
     }
 
     public static int getTypeCodeFromGbCode(String deviceId) {

src/main/java/com/genersoft/iot/vmp/media/zlm/ZLMHttpHookListener.java

@@ -9,9 +9,9 @@ import com.genersoft.iot.vmp.gb28181.bean.*;
 import com.genersoft.iot.vmp.gb28181.event.EventPublisher;
 import com.genersoft.iot.vmp.gb28181.session.AudioBroadcastManager;
 import com.genersoft.iot.vmp.gb28181.session.VideoStreamSessionManager;
-import com.genersoft.iot.vmp.gb28181.transmit.cmd.ISIPCommanderForPlatform;
 import com.genersoft.iot.vmp.gb28181.transmit.callback.DeferredResultHolder;
 import com.genersoft.iot.vmp.gb28181.transmit.callback.RequestMessage;
+import com.genersoft.iot.vmp.gb28181.transmit.cmd.ISIPCommanderForPlatform;
 import com.genersoft.iot.vmp.gb28181.transmit.cmd.impl.SIPCommander;
 import com.genersoft.iot.vmp.media.zlm.dto.HookType;
 import com.genersoft.iot.vmp.media.zlm.dto.MediaServerItem;
@@ -249,6 +249,7 @@ public class ZLMHttpHookListener {
             String channelId = ssrcTransactionForAll.get(0).getChannelId();
             DeviceChannel deviceChannel = storager.queryChannel(deviceId, channelId);
             if (deviceChannel != null) {
+
                 result.setEnable_audio(deviceChannel.isHasAudio());
             }
             // 如果是录像下载就设置视频间隔十秒
@@ -257,6 +258,11 @@ public class ZLMHttpHookListener {
                 result.setEnable_audio(true);
                 result.setEnable_mp4(true);
             }
+            // 如果是talk对讲，则默认获取声音
+            if (ssrcTransactionForAll.get(0).getType() == VideoStreamSessionManager.SessionType.talk) {
+                result.setEnable_audio(true);
+            }
+
         }
         return result;
     }
@@ -359,62 +365,30 @@ public class ZLMHttpHookListener {
 				}
 			}else if ("talk".equals(param.getApp())){
 				// 语音对讲推流  stream需要满足格式deviceId_channelId
-				if (param.isRegist() && param.getStream().indexOf("_") > 0) {
+                if (param.getStream().indexOf("_") > 0) {
-					String[] streamArray = param.getStream().split("_");
+                    String[] streamArray = param.getStream().split("_");
-					if (streamArray.length == 2) {
+                    if (streamArray.length == 2) {
-						String deviceId = streamArray[0];
+                        String deviceId = streamArray[0];
-						String channelId = streamArray[1];
+                        String channelId = streamArray[1];
-						Device device = deviceService.getDevice(deviceId);
+                        Device device = deviceService.getDevice(deviceId);
-						if (device != null) {
+                        if (device != null) {
-							DeviceChannel deviceChannel = storager.queryChannel(deviceId, channelId);
+                            if (param.isRegist()) {
-							if (deviceChannel != null) {
+                                if (audioBroadcastManager.exit(deviceId, channelId)) {
-								if (audioBroadcastManager.exit(deviceId, channelId)) {
+                                    playService.stopAudioBroadcast(deviceId, channelId);
-									// 直接推流
+                                }
-									SendRtpItem sendRtpItem =  redisCatchStorage.querySendRTPServer(null, null, param.getStream(), null);
+                                // 开启语音对讲通道
-									if (sendRtpItem == null) {
+                                playService.talkCmd(device, channelId, mediaInfo, param.getStream(), (msg)->{
-										// TODO 可能数据错误，重新开启语音通道
+                                    logger.info("[语音对讲] 通道建立成功, device: {}, channel: {}", deviceId, channelId);
-									}else {
+                                });
-										MediaServerItem mediaServerItem = mediaServerService.getOne(sendRtpItem.getMediaServerId());
+                            }else {
-										logger.info("rtp/{}开始向上级推流, 目标={}:{}，SSRC={}", sendRtpItem.getStreamId(), sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc());
+                                // 流注销
-										Map<String, Object> sendParam = new HashMap<>(12);
+                                playService.stopTalk(device, channelId, param.isRegist());
-										sendParam.put("vhost","__defaultVhost__");
+                            }
-										sendParam.put("app",sendRtpItem.getApp());
+                        } else{
-										sendParam.put("stream",sendRtpItem.getStreamId());
+                            logger.info("[语音对讲] 未找到设备：{}", deviceId);
-										sendParam.put("ssrc", sendRtpItem.getSsrc());
+                        }
-										sendParam.put("src_port", sendRtpItem.getLocalPort());
+                    }
-										sendParam.put("pt", sendRtpItem.getPt());
+                }
-										sendParam.put("use_ps", sendRtpItem.isUsePs() ? "1" : "0");
-										sendParam.put("only_audio", sendRtpItem.isOnlyAudio() ? "1" : "0");
-
-										JSONObject jsonObject;
-										if (sendRtpItem.isTcpActive()) {
-											jsonObject = zlmrtpServerFactory.startSendRtpPassive(mediaServerItem, sendParam);
-										} else {
-											sendParam.put("is_udp", sendRtpItem.isTcp() ? "0" : "1");
-											sendParam.put("dst_url", sendRtpItem.getIp());
-											sendParam.put("dst_port", sendRtpItem.getPort());
-											jsonObject = zlmrtpServerFactory.startSendRtpStream(mediaServerItem, sendParam);
-										}
-										if (jsonObject != null && jsonObject.getInteger("code") == 0) {
-											logger.info("[语音对讲] 自动推流成功, device: {}, channel: {}", deviceId, channelId);
-										}
-									}
-								}else {
-									// 开启语音对讲通道
-									MediaServerItem mediaServerItem = mediaServerService.getOne(param.getMediaServerId());
-									playService.talk(mediaServerItem, device, channelId, (mediaServer, jsonObject)->{
-										System.out.println("开始推流");
-									}, eventResult -> {
-										System.out.println(eventResult.msg);
-									}, ()->{
-										System.out.println("超时");
-									});
-								}
-
-							}
-						}
-					}
-				}
 
 			}else{
 				if (!"rtp".equals(param.getApp())){
@@ -474,16 +448,21 @@ public class ZLMHttpHookListener {
                                 ParentPlatform platform = storager.queryParentPlatByServerGBId(platformId);
                                 Device device = deviceService.getDevice(platformId);
 
-                                try {
+
                                     if (platform != null) {
-                                        commanderFroPlatform.streamByeCmd(platform, sendRtpItem);
+                                        try {
+                                            commanderFroPlatform.streamByeCmd(platform, sendRtpItem);
+                                        } catch (SipException | InvalidArgumentException | ParseException e) {
+                                            logger.error("[命令发送失败] 国标级联 发送BYE: {}", e.getMessage());
+                                        }
                                     } else {
-                                        cmder.streamByeCmd(device, sendRtpItem.getChannelId(), param.getStream(), sendRtpItem.getCallId());
+                                        try {
+                                            cmder.streamByeCmd(device, sendRtpItem.getChannelId(), param.getStream(), sendRtpItem.getCallId());
+                                        } catch (SipException | InvalidArgumentException | ParseException |
+                                                 SsrcTransactionNotFoundException e) {
+                                            logger.error("[命令发送失败] 发送BYE: {}", e.getMessage());
+                                        }
                                     }
-                                } catch (SipException | InvalidArgumentException | ParseException |
-                                         SsrcTransactionNotFoundException e) {
-                                    logger.error("[命令发送失败] 国标级联 发送BYE: {}", e.getMessage());
-                                }
                             }
                         }
                     }
@@ -526,7 +505,7 @@ public class ZLMHttpHookListener {
 								logger.error("[命令发送失败] 国标级联 发送BYE: {}", e.getMessage());
 							}
 							redisCatchStorage.deleteSendRTPServer(parentPlatform.getServerGBId(), sendRtpItem.getChannelId(),
-									sendRtpItem.getCallId(), sendRtpItem.getStreamId());
+									sendRtpItem.getCallId(), sendRtpItem.getStream());
 						}
 					}
 				}
@@ -555,8 +534,7 @@ public class ZLMHttpHookListener {
                         try {
                             cmder.streamByeCmd(device, streamInfoForPlayBackCatch.getChannelId(),
                                     streamInfoForPlayBackCatch.getStream(), null);
-                        } catch (InvalidArgumentException | ParseException | SipException |
+                        } catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
-                                 SsrcTransactionNotFoundException e) {
                             logger.error("[无人观看]回放， 发送BYE失败 {}", e.getMessage());
                         }
                     }
@@ -572,6 +550,13 @@ public class ZLMHttpHookListener {
                 ret.put("close", false);
                 return ret;
             }
+            SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(null, null, param.getStream(), null);
+            if ("talk".equals(sendRtpItem.getApp())){
+                ret.put("close", false);
+                return ret;
+            }
+        }else if ("talk".equals(param.getApp()) || "broadcast".equals(param.getApp())){
+            ret.put("close", false);
         } else {
             // 非国标流 推流/拉流代理
             // 拉流代理
@@ -733,7 +718,7 @@ public class ZLMHttpHookListener {
                         logger.error("[命令发送失败] 国标级联 发送BYE: {}", e.getMessage());
                     }
                     redisCatchStorage.deleteSendRTPServer(parentPlatform.getServerGBId(), sendRtpItem.getChannelId(),
-                            sendRtpItem.getCallId(), sendRtpItem.getStreamId());
+                            sendRtpItem.getCallId(), sendRtpItem.getStream());
                 }
             }
         });

src/main/java/com/genersoft/iot/vmp/media/zlm/ZLMRESTfulUtils.java

@@ -291,6 +291,10 @@ public class ZLMRESTfulUtils {
         return sendPost(mediaServerItem, "startSendRtpPassive",param, null);
     }
 
+    public JSONObject startSendRtpPassive(MediaServerItem mediaServerItem, Map<String, Object> param, RequestCallback callback) {
+        return sendPost(mediaServerItem, "startSendRtpPassive",param, callback);
+    }
+
     public JSONObject stopSendRtp(MediaServerItem mediaServerItem, Map<String, Object> param) {
         return sendPost(mediaServerItem, "stopSendRtp",param, null);
     }

src/main/java/com/genersoft/iot/vmp/media/zlm/ZLMRTPServerFactory.java

@@ -229,7 +229,7 @@ public class ZLMRTPServerFactory {
         sendRtpItem.setPort(port);
         sendRtpItem.setSsrc(ssrc);
         sendRtpItem.setApp(app);
-        sendRtpItem.setStreamId(stream);
+        sendRtpItem.setStream(stream);
         sendRtpItem.setPlatformId(platformId);
         sendRtpItem.setChannelId(channelId);
         sendRtpItem.setTcp(tcp);
@@ -290,6 +290,10 @@ public class ZLMRTPServerFactory {
         return zlmresTfulUtils.startSendRtpPassive(mediaServerItem, param);
     }
 
+    public JSONObject startSendRtpPassive(MediaServerItem mediaServerItem, Map<String, Object>param, ZLMRESTfulUtils.RequestCallback callback) {
+        return zlmresTfulUtils.startSendRtpPassive(mediaServerItem, param, callback);
+    }
+
     /**
      * 查询待转推的流是否就绪
      */
@@ -343,7 +347,7 @@ public class ZLMRTPServerFactory {
             result= true;
             logger.info("[停止RTP推流] 成功");
         } else {
-            logger.error("[停止RTP推流] 失败: {}, 参数：{}->\r\n{}",jsonObject.getString("msg"), JSON.toJSON(param), jsonObject);
+            logger.warn("[停止RTP推流] 失败: {}, 参数：{}->\r\n{}",jsonObject.getString("msg"), JSON.toJSON(param), jsonObject);
         }
         return result;
     }

src/main/java/com/genersoft/iot/vmp/service/IPlayService.java

@@ -12,6 +12,7 @@ import com.genersoft.iot.vmp.service.bean.PlayBackCallback;
 import com.genersoft.iot.vmp.service.bean.SSRCInfo;
 import com.genersoft.iot.vmp.vmanager.bean.AudioBroadcastResult;
 import com.genersoft.iot.vmp.vmanager.gb28181.play.bean.AudioBroadcastEvent;
+import com.genersoft.iot.vmp.vmanager.gb28181.play.bean.AudioEvent;
 import gov.nist.javax.sip.message.SIPResponse;
 
 import javax.sip.InvalidArgumentException;
@@ -27,10 +28,6 @@ public interface IPlayService {
 
     void onPublishHandlerForPlay(MediaServerItem mediaServerItem, JSONObject resonse, String deviceId, String channelId);
 
-    void talk(MediaServerItem mediaServerItem, Device device, String channelId,
-              ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
-              Runnable timeoutCallback);
-
     void play(MediaServerItem mediaServerItem, SSRCInfo ssrcInfo, Device device, String channelId,
               ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
               InviteTimeOutCallback timeoutCallback);
@@ -57,7 +54,7 @@ public interface IPlayService {
 
     void zlmServerOnline(String mediaServerId);
 
-    AudioBroadcastResult audioBroadcast(Device device, String channelId);
+    AudioBroadcastResult audioBroadcast(Device device, String channelId, Boolean broadcastMode);
     void stopAudioBroadcast(String deviceId, String channelId);
 
     void audioBroadcastCmd(Device device, String channelId, int timeout, MediaServerItem mediaServerItem, String sourceApp, String sourceStream, AudioBroadcastEvent event) throws InvalidArgumentException, ParseException, SipException;
@@ -70,4 +67,8 @@ public interface IPlayService {
 
     void startSendRtpStreamHand(SendRtpItem sendRtpItem, ParentPlatform parentPlatform,
                                 JSONObject jsonObject, Map<String, Object> param, CallIdHeader callIdHeader);
+
+    void talkCmd(Device device, String channelId, MediaServerItem mediaServerItem, String stream, AudioEvent event);
+
+    void stopTalk(Device device, String channelId, Boolean streamIsReady);
 }

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

@@ -202,7 +202,7 @@ public class DeviceServiceImpl implements IDeviceService {
                     Map<String, Object> param = new HashMap<>();
                     param.put("vhost", "__defaultVhost__");
                     param.put("app", sendRtpItem.getApp());
-                    param.put("stream", sendRtpItem.getStreamId());
+                    param.put("stream", sendRtpItem.getStream());
                     zlmresTfulUtils.stopSendRtp(mediaInfo, param);
                 }
 

src/main/java/com/genersoft/iot/vmp/service/impl/PlatformServiceImpl.java

@@ -253,7 +253,7 @@ public class PlatformServiceImpl implements IPlatformService {
                 Map<String, Object> param = new HashMap<>(3);
                 param.put("vhost", "__defaultVhost__");
                 param.put("app", sendRtpItem.getApp());
-                param.put("stream", sendRtpItem.getStreamId());
+                param.put("stream", sendRtpItem.getStream());
                 zlmrtpServerFactory.stopSendRtpStream(mediaInfo, param);
             }
         }

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

@@ -41,7 +41,7 @@ import com.genersoft.iot.vmp.vmanager.bean.AudioBroadcastResult;
 import com.genersoft.iot.vmp.vmanager.bean.ErrorCode;
 import com.genersoft.iot.vmp.vmanager.bean.StreamContent;
 import com.genersoft.iot.vmp.vmanager.bean.WVPResult;
-import com.genersoft.iot.vmp.vmanager.gb28181.play.bean.AudioBroadcastEvent;
+import com.genersoft.iot.vmp.vmanager.gb28181.play.bean.AudioEvent;
 import gov.nist.javax.sip.message.SIPResponse;
 import org.slf4j.Logger;
 import org.slf4j.LoggerFactory;
@@ -134,8 +134,8 @@ public class PlayServiceImpl implements IPlayService {
 
     @Override
     public void play(MediaServerItem mediaServerItem, String deviceId, String channelId,
-                                 ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
+                     ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
-                                 Runnable timeoutCallback) {
+                     Runnable timeoutCallback) {
         if (mediaServerItem == null) {
             throw new ControllerException(ErrorCode.ERROR100.getCode(), "未找到可用的zlm");
         }
@@ -243,194 +243,148 @@ public class PlayServiceImpl implements IPlayService {
         }
     }
 
-    @Override
+    private void talk(MediaServerItem mediaServerItem, Device device, String channelId, String stream,
-    public void talk(MediaServerItem mediaServerItem, Device device, String channelId,
+                      ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
-                     ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
+                      Runnable timeoutCallback, AudioEvent audioEvent) {
-                     Runnable timeoutCallback) {
+
-        String streamId = null;
+        String playSsrc = mediaServerItem.getSsrcConfig().getPlaySsrc();
-        if (mediaServerItem.isRtpEnable()) {
+        if (playSsrc == null) {
-            streamId = String.format("%s_%s", device.getDeviceId(), channelId);
+            audioEvent.call("ssrc已经用尽");
+            return;
         }
-        SSRCInfo ssrcInfo = mediaServerService.openRTPServer(mediaServerItem, streamId, device.isSsrcCheck(), false);
+        SendRtpItem sendRtpItem = new SendRtpItem();
-        logger.info("[对讲开始] deviceId: {}, channelId: {},收流端口： {}, 收流模式：{}, SSRC: {}, SSRC校验：{}", device.getDeviceId(), channelId, ssrcInfo.getPort(), device.getStreamMode(), ssrcInfo.getSsrc(), device.isSsrcCheck());
+        sendRtpItem.setApp("talk");
+        sendRtpItem.setStream(stream);
+        sendRtpItem.setSsrc(playSsrc);
+        sendRtpItem.setDeviceId(device.getDeviceId());
+        sendRtpItem.setPlatformId(device.getDeviceId());
+        sendRtpItem.setChannelId(channelId);
+        sendRtpItem.setRtcp(false);
+        sendRtpItem.setMediaServerId(mediaServerItem.getId());
+        sendRtpItem.setOnlyAudio(true);
+        sendRtpItem.setPlayType(InviteStreamType.TALK);
+        sendRtpItem.setPt(8);
+        sendRtpItem.setStatus(1);
+        sendRtpItem.setTcpActive(false);
+        sendRtpItem.setTcp(true);
+        sendRtpItem.setUsePs(false);
+        sendRtpItem.setReceiveStream(stream + "_talk");
+
+
+        int port = zlmrtpServerFactory.keepPort(mediaServerItem, playSsrc);
+        //端口获取失败的ssrcInfo 没有必要发送点播指令
+        if (port <= 0) {
+            logger.info("[语音对讲] 端口分配异常，deviceId={},channelId={}", device.getDeviceId(), channelId);
+            audioEvent.call("端口分配异常");
+            return;
+        }
+        sendRtpItem.setLocalPort(port);
+        sendRtpItem.setPort(port);
+        logger.info("[语音对讲]开始 deviceId: {}, channelId: {},收流端口： {}, 收流模式：{}, SSRC: {}, SSRC校验：{}", device.getDeviceId(), channelId, sendRtpItem.getLocalPort(), device.getStreamMode(), sendRtpItem.getSsrc(), false);
         // 超时处理
         String timeOutTaskKey = UUID.randomUUID().toString();
-        SSRCInfo finalSsrcInfo = ssrcInfo;
-        System.out.println("设置超时任务： " + timeOutTaskKey);
         dynamicTask.startDelay(timeOutTaskKey, () -> {
 
-            logger.info("[对讲超时] 收流超时 deviceId: {}, channelId: {}，端口：{}, SSRC: {}", device.getDeviceId(), channelId, finalSsrcInfo.getPort(), finalSsrcInfo.getSsrc());
+            logger.info("[语音对讲] 收流超时 deviceId: {}, channelId: {}，端口：{}, SSRC: {}", device.getDeviceId(), channelId, sendRtpItem.getPort(), sendRtpItem.getSsrc());
             timeoutCallback.run();
             // 点播超时回复BYE 同时释放ssrc以及此次点播的资源
             try {
-                cmder.streamByeCmd(device, channelId, finalSsrcInfo.getStream(), null);
+                cmder.streamByeCmd(device, channelId, sendRtpItem.getStream(), null);
-            } catch (InvalidArgumentException | ParseException | SipException e) {
+            } catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
-                logger.error("[对讲超时]， 发送BYE失败 {}", e.getMessage());
+                logger.error("[语音对讲]超时， 发送BYE失败 {}", e.getMessage());
-            } catch (SsrcTransactionNotFoundException e) {
+            } finally {
                 timeoutCallback.run();
-                mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
+                mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
-                mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
+                streamSession.remove(device.getDeviceId(), channelId, sendRtpItem.getStream());
-                streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
             }
         }, userSetting.getPlayTimeout());
-        final String ssrc = ssrcInfo.getSsrc();
-        final String stream = ssrcInfo.getStream();
-        //端口获取失败的ssrcInfo 没有必要发送点播指令
-        if (ssrcInfo.getPort() <= 0) {
-            logger.info("[对讲] 端口分配异常，deviceId={},channelId={},ssrcInfo={}", device.getDeviceId(), channelId, ssrcInfo);
-            return;
-        }
 
         String callId = SipUtils.getNewCallId();
-        boolean pushing = false;
+
+        zlmrtpServerFactory.releasePort(mediaServerItem, playSsrc);
+        Map<String, Object> param = new HashMap<>(12);
+        param.put("vhost","__defaultVhost__");
+        param.put("app", sendRtpItem.getApp());
+        param.put("stream", sendRtpItem.getStream());
+        param.put("ssrc", sendRtpItem.getSsrc());
+        param.put("src_port", sendRtpItem.getLocalPort());
+        param.put("pt", sendRtpItem.getPt());
+        param.put("use_ps", sendRtpItem.isUsePs() ? "1" : "0");
+        param.put("only_audio", sendRtpItem.isOnlyAudio() ? "1" : "0");
+        param.put("is_udp", sendRtpItem.isTcp() ? "0" : "1");
+        param.put("recv_stream_id", sendRtpItem.getReceiveStream());
+        param.put("close_delay_ms", userSetting.getPlayTimeout() * 1000);
+
+        zlmrtpServerFactory.startSendRtpPassive(mediaServerItem, param, jsonObject -> {
+            if (jsonObject == null || jsonObject.getInteger("code") != 0 ) {
+                mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
+                logger.info("[语音对讲]失败 deviceId: {}, channelId: {}", device.getDeviceId(), channelId);
+                audioEvent.call("失败, " + jsonObject.getString("msg"));
+                // 查看是否已经建立了通道，存在则发送bye
+                stopTalk(device, channelId);
+            }
+        });
+
+
         // 查看设备是否已经在推流
-//        MediaItem mediaItem = zlmrtpServerFactory.getMediaInfo(mediaServerItem, "rtp",ssrcInfo.getStream());
+        try {
-//        if (mediaItem != null) {
+            cmder.talkStreamCmd(mediaServerItem, sendRtpItem, device, channelId, callId, (MediaServerItem mediaServerItemInuse, JSONObject response) -> {
-//            SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem,
+                logger.info("[语音对讲] 流已生成， 开始推流： " + response.toJSONString());
-//                    mediaItem.getOriginSock().getPeer_ip(), mediaItem.getOriginSock().getPeer_port(), ssrcInfo.getSsrc(), device.getDeviceId(),
+                dynamicTask.stop(timeOutTaskKey);
-//                    device.getDeviceId(), channelId,
+                // TODO 暂不做处理
-//                    false);
+            }, (MediaServerItem mediaServerItemInuse, JSONObject json) -> {
-//
+                logger.info("[语音对讲] 设备开始推流： " + json.toJSONString());
-//            sendRtpItem.setTcpActive(false);
+                dynamicTask.stop(timeOutTaskKey);
-//            sendRtpItem.setCallId(callId);
-//            sendRtpItem.setPlayType(InviteStreamType.TALK);
-//            sendRtpItem.setStatus(1);
-//            sendRtpItem.setIp(mediaItem.getOriginSock().getPeer_ip());
-//            sendRtpItem.setPort(mediaItem.getOriginSock().getPeer_port());
-//            sendRtpItem.setTcpActive(false);
-//            sendRtpItem.setStreamId(ssrcInfo.getStream());
-//            sendRtpItem.setApp("1000");
-//            sendRtpItem.setStreamId("1000");
-//            sendRtpItem.setSsrc(ssrc);
-//            sendRtpItem.setOnlyAudio(true);
-//            redisCatchStorage.updateSendRTPSever(sendRtpItem);
-//
-//            Map<String, Object> param = new HashMap<>(12);
-//            param.put("vhost","__defaultVhost__");
-//            param.put("app",sendRtpItem.getApp());
-//            param.put("stream",sendRtpItem.getStreamId());
-//            param.put("ssrc", sendRtpItem.getSsrc());
-//            param.put("dst_url", sendRtpItem.getIp());
-//            param.put("dst_port", sendRtpItem.getPort());
-//            param.put("src_port", sendRtpItem.getLocalPort());
-//            param.put("pt", sendRtpItem.getPt());
-//            param.put("use_ps", sendRtpItem.isUsePs() ? "1" : "0");
-//            param.put("is_udp", sendRtpItem.isTcp() ? "0" : "1");
-//            param.put("only_audio", sendRtpItem.isOnlyAudio() ? "1" : "0");
-//            JSONObject jsonObject = zlmrtpServerFactory.startSendRtpStream(mediaServerItem, param);
-//            System.out.println(2222);
-//            System.out.println(jsonObject);
-//        }else {
-            try {
-                cmder.talkStreamCmd(mediaServerItem, ssrcInfo, device, channelId, callId, (MediaServerItem mediaServerItemInuse, JSONObject response) -> {
-                    logger.info("[对讲] 流已生成， 开始推流： " + response.toJSONString());
-                    dynamicTask.stop(timeOutTaskKey);
-                    // TODO 暂不做处理
-                }, (MediaServerItem mediaServerItemInuse, JSONObject json) -> {
-                    logger.info("[对讲] 设备开始推流： " + json.toJSONString());
-                    dynamicTask.stop(timeOutTaskKey);
-                    // 获取远程IP端口 作为回复语音流的地址
-                    String ip = json.getString("ip");
-                    Integer port = json.getInteger("port");
-                    logger.info("[设备开始推流]{}/{}, 来自ip：{}, 端口：{}", device.getDeviceId(), channelId, ip, port);
-                    // 查看平台推流是否就绪
-//                    Boolean ready = zlmrtpServerFactory.isStreamReady(mediaServerItemInuse, "talk", stream);
-//                    if (!ready) {
-//                        try {
-//                            cmder.streamByeCmd(device, channelId, finalSsrcInfo.getStream(), null);
-//                        } catch (InvalidArgumentException | ParseException | SipException e) {
-//                            logger.error("[对讲超时]， 发送BYE失败 {}", e.getMessage());
-//                        } catch (SsrcTransactionNotFoundException e) {
-//                            timeoutCallback.run();
-//                            mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
-//                            mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
-//                            streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
-//                        }
-//                    }else {
-//                        try {
-//                            Thread.sleep(1000);
-//                        } catch (InterruptedException e) {
-//                            throw new RuntimeException(e);
-//                        }
-                        SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, ip, port, ssrcInfo.getSsrc(), device.getDeviceId(),
-                                device.getDeviceId(), channelId,
-                                false, false);
 
+            }, (event) -> {
+                dynamicTask.stop(timeOutTaskKey);
 
-//                        if (sendRtpItem.getLocalPort() == 0) {
+                if (event.event instanceof ResponseEvent) {
-//                            logger.warn("服务器端口资源不足");
+                    ResponseEvent responseEvent = (ResponseEvent) event.event;
-//                            try {
+                    if (responseEvent.getResponse() instanceof SIPResponse) {
-//                                cmder.streamByeCmd(device, channelId, finalSsrcInfo.getStream(), null);
+                        SIPResponse response = (SIPResponse) responseEvent.getResponse();
-//                            } catch (InvalidArgumentException | ParseException | SipException e) {
+                        sendRtpItem.setFromTag(response.getFromTag());
-//                                logger.error("[对讲超时]， 发送BYE失败 {}", e.getMessage());
+                        sendRtpItem.setToTag(response.getToTag());
-//                            } catch (SsrcTransactionNotFoundException e) {
+                        sendRtpItem.setCallId(response.getCallIdHeader().getCallId());
-//                                timeoutCallback.run();
-//                                mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
-//                                mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
-//                                streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
-//                            }
-//                            return;
-//                        }
-                        sendRtpItem.setTcpActive(false);
-                        sendRtpItem.setCallId(callId);
-                        sendRtpItem.setPlayType(InviteStreamType.TALK);
-                        sendRtpItem.setStatus(1);
-                        sendRtpItem.setIp(ip);
-                        sendRtpItem.setPort(port);
-                        sendRtpItem.setTcpActive(false);
-                        sendRtpItem.setApp("1000");
-                        sendRtpItem.setStreamId("1000");
-                        sendRtpItem.setSsrc(ssrc);
-                        sendRtpItem.setOnlyAudio(true);
-                        sendRtpItem.setRtcp(false);
                         redisCatchStorage.updateSendRTPSever(sendRtpItem);
 
-                        Map<String, Object> param = new HashMap<>(12);
+                        streamSession.put(device.getDeviceId(), channelId, "talk",
-                        param.put("vhost","__defaultVhost__");
+                                sendRtpItem.getStream(), sendRtpItem.getSsrc(), sendRtpItem.getMediaServerId(),
-                        param.put("app",sendRtpItem.getApp());
+                                response, VideoStreamSessionManager.SessionType.talk);
-                        param.put("stream",sendRtpItem.getStreamId());
+                    } else {
-                        param.put("ssrc", sendRtpItem.getSsrc());
+                        logger.error("[语音对讲]收到的消息错误，response不是SIPResponse");
-                        param.put("dst_url", sendRtpItem.getIp());
+                    }
-                        param.put("dst_port", sendRtpItem.getPort());
+                } else {
-                        param.put("src_port", sendRtpItem.getLocalPort());
+                    logger.error("[语音对讲]收到的消息错误，event不是ResponseEvent");
-                        param.put("pt", sendRtpItem.getPt());
+                }
-                        param.put("use_ps", sendRtpItem.isUsePs() ? "1" : "0");
-                        param.put("is_udp", sendRtpItem.isTcp() ? "0" : "1");
-                        param.put("only_audio", sendRtpItem.isOnlyAudio() ? "1" : "0");
-                        JSONObject jsonObject = zlmrtpServerFactory.startSendRtpStream(mediaServerItemInuse, param);
-                        System.out.println(11111);
-                        System.out.println(sendRtpItem.getIp() + ":" + sendRtpItem.getPort());
-//                        System.out.println(jsonObject);
-//                    }
 
-                }, (event) -> {
+            }, (event) -> {
-
-                }, (event) -> {
-                    dynamicTask.stop(timeOutTaskKey);
-                    mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
-                    // 释放ssrc
-                    mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
-
-                    streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
-                    errorEvent.response(event);
-                });
-            } catch (InvalidArgumentException | SipException | ParseException e) {
-
-                logger.error("[命令发送失败] 对讲消息: {}", e.getMessage());
                 dynamicTask.stop(timeOutTaskKey);
-                mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
+                mediaServerService.closeRTPServer(mediaServerItem, sendRtpItem.getStream());
                 // 释放ssrc
-                mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
+                mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
+                streamSession.remove(device.getDeviceId(), channelId, sendRtpItem.getStream());
+                errorEvent.response(event);
+            });
+        } catch (InvalidArgumentException | SipException | ParseException e) {
 
-                streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
+            logger.error("[命令发送失败] 对讲消息: {}", e.getMessage());
-                SipSubscribe.EventResult eventResult = new SipSubscribe.EventResult(new CmdSendFailEvent(null));
+            dynamicTask.stop(timeOutTaskKey);
-                eventResult.msg = "命令发送失败";
+            mediaServerService.closeRTPServer(mediaServerItem, sendRtpItem.getStream());
-                errorEvent.response(eventResult);
+            // 释放ssrc
-            }
+            mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
+
+            streamSession.remove(device.getDeviceId(), channelId, sendRtpItem.getStream());
+            SipSubscribe.EventResult eventResult = new SipSubscribe.EventResult(new CmdSendFailEvent(null));
+            eventResult.msg = "命令发送失败";
+            errorEvent.response(eventResult);
+        }
 //        }
 
     }
 
+
+
     @Override
     public void play(MediaServerItem mediaServerItem, SSRCInfo ssrcInfo, Device device, String channelId,
                      ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
@@ -446,7 +400,8 @@ public class PlayServiceImpl implements IPlayService {
                 // 点播超时回复BYE 同时释放ssrc以及此次点播的资源
                 try {
                     cmder.streamByeCmd(device, channelId, ssrcInfo.getStream(), null);
-                } catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
+                } catch (InvalidArgumentException | ParseException | SipException |
+                         SsrcTransactionNotFoundException e) {
                     logger.error("[点播超时]， 发送BYE失败 {}", e.getMessage());
                 } finally {
                     timeoutCallback.run(1, "收流超时");
@@ -483,7 +438,7 @@ public class PlayServiceImpl implements IPlayService {
                 onPublishHandlerForPlay(mediaServerItemInuse, response, device.getDeviceId(), channelId);
                 hookEvent.response(mediaServerItemInuse, response);
                 logger.info("[点播成功] deviceId: {}, channelId: {}", device.getDeviceId(), channelId);
-                String streamUrl = String.format("http://127.0.0.1:%s/%s/%s.live.flv", mediaServerItemInuse.getHttpPort(), "rtp",  ssrcInfo.getStream());
+                String streamUrl = String.format("http://127.0.0.1:%s/%s/%s.live.flv", mediaServerItemInuse.getHttpPort(), "rtp", ssrcInfo.getStream());
                 String path = "snap";
                 String fileName = device.getDeviceId() + "_" + channelId + ".jpg";
                 // 请求截图
@@ -652,8 +607,8 @@ public class PlayServiceImpl implements IPlayService {
 
     @Override
     public void playBack(String deviceId, String channelId, String startTime,
-                                                          String endTime, InviteStreamCallback inviteStreamCallback,
+                         String endTime, InviteStreamCallback inviteStreamCallback,
-                                                          PlayBackCallback callback) {
+                         PlayBackCallback callback) {
         Device device = storager.queryVideoDevice(deviceId);
         if (device == null) {
             return;
@@ -666,9 +621,9 @@ public class PlayServiceImpl implements IPlayService {
 
     @Override
     public void playBack(MediaServerItem mediaServerItem, SSRCInfo ssrcInfo,
-                                                          String deviceId, String channelId, String startTime,
+                         String deviceId, String channelId, String startTime,
-                                                          String endTime, InviteStreamCallback infoCallBack,
+                         String endTime, InviteStreamCallback infoCallBack,
-                                                          PlayBackCallback playBackCallback) {
+                         PlayBackCallback playBackCallback) {
         if (mediaServerItem == null || ssrcInfo == null) {
             return;
         }
@@ -792,7 +747,6 @@ public class PlayServiceImpl implements IPlayService {
     }
 
 
-
     @Override
     public void download(String deviceId, String channelId, String startTime, String endTime, int downloadSpeed, InviteStreamCallback infoCallBack, PlayBackCallback playBackCallback) {
         Device device = storager.queryVideoDevice(deviceId);
@@ -977,7 +931,7 @@ public class PlayServiceImpl implements IPlayService {
                         cmder.streamByeCmd(device, ssrcTransaction.getChannelId(),
                                 ssrcTransaction.getStream(), null);
                     } catch (InvalidArgumentException | ParseException | SipException |
-                            SsrcTransactionNotFoundException e) {
+                             SsrcTransactionNotFoundException e) {
                         logger.error("[zlm离线]为正在使用此zlm的设备， 发送BYE失败 {}", e.getMessage());
                     }
                 }
@@ -986,7 +940,8 @@ public class PlayServiceImpl implements IPlayService {
     }
 
     @Override
-    public AudioBroadcastResult audioBroadcast(Device device, String channelId) {
+    public AudioBroadcastResult audioBroadcast(Device device, String channelId, Boolean broadcastMode) {
+        // TODO 必须多端口模式才支持语音喊话鹤语音对讲
         if (device == null || channelId == null) {
             return null;
         }
@@ -997,15 +952,15 @@ public class PlayServiceImpl implements IPlayService {
             return null;
         }
         MediaServerItem mediaServerItem = mediaServerService.getMediaServerForMinimumLoad(null);
-        String app = "broadcast";
+        if (broadcastMode == null) {
-        // TODO 从sip user agent中判断是什么品牌设备，大华默认使用talk模式，其他使用broadcast模式
+            broadcastMode = true;
-//        String app = "talk";
+        }
+        String app = broadcastMode?"broadcast":"talk";
         String stream = device.getDeviceId() + "_" + channelId;
-        StreamInfo broadcast = mediaService.getStreamInfoByAppAndStream(mediaServerItem, "broadcast", stream, null, null, null, false);
         AudioBroadcastResult audioBroadcastResult = new AudioBroadcastResult();
         audioBroadcastResult.setApp(app);
         audioBroadcastResult.setStream(stream);
-        audioBroadcastResult.setStreamInfo(new StreamContent(mediaService.getStreamInfoByAppAndStream(mediaServerItem, app, stream, null, null, null,false)));
+        audioBroadcastResult.setStreamInfo(new StreamContent(mediaService.getStreamInfoByAppAndStream(mediaServerItem, app, stream, null, null, null, false)));
         audioBroadcastResult.setCodec("G.711");
         return audioBroadcastResult;
     }
@@ -1037,6 +992,18 @@ public class PlayServiceImpl implements IPlayService {
                 }
             }
         }
+        SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(device.getDeviceId(), channelId, null, null);
+        if (sendRtpItem != null) {
+            MediaServerItem mediaServer = mediaServerService.getOne(sendRtpItem.getMediaServerId());
+            Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServer, "rtp", sendRtpItem.getReceiveStream());
+            if (streamReady) {
+                logger.warn("[语音对讲] 进行中： {}", channelId);
+                event.call("语音对讲进行中");
+                return;
+            } else {
+                stopTalk(device, channelId);
+            }
+        }
 
         // 发送通知
         cmder.audioBroadcastCmd(device, channelId, eventResultForOk -> {
@@ -1053,19 +1020,18 @@ public class PlayServiceImpl implements IPlayService {
     }
 
 
-
     @Override
     public void stopAudioBroadcast(String deviceId, String channelId) {
         List<AudioBroadcastCatch> audioBroadcastCatchList = new ArrayList<>();
         if (channelId == null) {
             audioBroadcastCatchList.addAll(audioBroadcastManager.get(deviceId));
-        }else {
+        } else {
             audioBroadcastCatchList.add(audioBroadcastManager.get(deviceId, channelId));
         }
         if (audioBroadcastCatchList.size() > 0) {
             for (AudioBroadcastCatch audioBroadcastCatch : audioBroadcastCatchList) {
                 Device device = deviceService.getDevice(deviceId);
-                if (device == null || audioBroadcastCatch == null ) {
+                if (device == null || audioBroadcastCatch == null) {
                     return;
                 }
                 SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(deviceId, audioBroadcastCatch.getChannelId(), null, null);
@@ -1075,7 +1041,7 @@ public class PlayServiceImpl implements IPlayService {
                     Map<String, Object> param = new HashMap<>();
                     param.put("vhost", "__defaultVhost__");
                     param.put("app", sendRtpItem.getApp());
-                    param.put("stream", sendRtpItem.getStreamId());
+                    param.put("stream", sendRtpItem.getStream());
                     zlmresTfulUtils.stopSendRtp(mediaInfo, param);
                     try {
                         cmder.streamByeCmd(device, sendRtpItem.getChannelId(), audioBroadcastCatch.getSipTransactionInfo(), null);
@@ -1199,12 +1165,12 @@ public class PlayServiceImpl implements IPlayService {
 
         String is_Udp = sendRtpItem.isTcp() ? "0" : "1";
         MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
-        logger.info("收到ACK，rtp/{}开始向上级推流, 目标={}:{}，SSRC={}, RTCP={}", sendRtpItem.getStreamId(),
+        logger.info("收到ACK，rtp/{}开始向上级推流, 目标={}:{}，SSRC={}, RTCP={}", sendRtpItem.getStream(),
                 sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isRtcp());
         Map<String, Object> param = new HashMap<>(12);
-        param.put("vhost","__defaultVhost__");
+        param.put("vhost", "__defaultVhost__");
-        param.put("app",sendRtpItem.getApp());
+        param.put("app", sendRtpItem.getApp());
-        param.put("stream",sendRtpItem.getStreamId());
+        param.put("stream", sendRtpItem.getStream());
         param.put("ssrc", sendRtpItem.getSsrc());
         param.put("src_port", sendRtpItem.getLocalPort());
         param.put("pt", sendRtpItem.getPt());
@@ -1213,12 +1179,12 @@ public class PlayServiceImpl implements IPlayService {
         param.put("is_udp", is_Udp);
         if (!sendRtpItem.isTcp()) {
             // udp模式下开启rtcp保活
-            param.put("udp_rtcp_timeout", sendRtpItem.isRtcp()? "1":"0");
+            param.put("udp_rtcp_timeout", sendRtpItem.isRtcp() ? "1" : "0");
         }
 
         if (mediaInfo == null) {
             RequestPushStreamMsg requestPushStreamMsg = RequestPushStreamMsg.getInstance(
-                    sendRtpItem.getMediaServerId(), sendRtpItem.getApp(), sendRtpItem.getStreamId(),
+                    sendRtpItem.getMediaServerId(), sendRtpItem.getApp(), sendRtpItem.getStream(),
                     sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isTcp(),
                     sendRtpItem.getLocalPort(), sendRtpItem.getPt(), sendRtpItem.isUsePs(), sendRtpItem.isOnlyAudio());
             redisGbPlayMsgListener.sendMsgForStartSendRtpStream(sendRtpItem.getServerId(), requestPushStreamMsg, json -> {
@@ -1233,16 +1199,16 @@ public class PlayServiceImpl implements IPlayService {
                 if (zlmrtpServerFactory.releasePort(mediaInfo, sendRtpItem.getSsrc())) {
                     if (sendRtpItem.isTcpActive()) {
                         startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpPassive(mediaInfo, param);
-                    }else {
+                    } else {
                         param.put("dst_url", sendRtpItem.getIp());
                         param.put("dst_port", sendRtpItem.getPort());
                         startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpStream(mediaInfo, param);
                     }
                 }
-            }else {
+            } else {
                 if (sendRtpItem.isTcpActive()) {
                     startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpPassive(mediaInfo, param);
-                }else {
+                } else {
                     param.put("dst_url", sendRtpItem.getIp());
                     param.put("dst_port", sendRtpItem.getPort());
                     startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpStream(mediaInfo, param);
@@ -1260,10 +1226,10 @@ public class PlayServiceImpl implements IPlayService {
         if (jsonObject == null) {
             logger.error("RTP推流失败: 请检查ZLM服务");
         } else if (jsonObject.getInteger("code") == 0) {
-            logger.info("调用ZLM推流接口, 结果： {}",  jsonObject);
+            logger.info("调用ZLM推流接口, 结果： {}", jsonObject);
-            logger.info("RTP推流成功[ {}/{} ]，{}->{}:{}, " ,param.get("app"), param.get("stream"), jsonObject.getString("local_port"), param.get("dst_url"), param.get("dst_port"));
+            logger.info("RTP推流成功[ {}/{} ]，{}->{}:{}, ", param.get("app"), param.get("stream"), jsonObject.getString("local_port"), param.get("dst_url"), param.get("dst_port"));
         } else {
-            logger.error("RTP推流失败: {}, 参数：{}",jsonObject.getString("msg"), JSON.toJSONString(param));
+            logger.error("RTP推流失败: {}, 参数：{}", jsonObject.getString("msg"), JSON.toJSONString(param));
             if (sendRtpItem.isOnlyAudio()) {
                 Device device = deviceService.getDevice(sendRtpItem.getDeviceId());
                 AudioBroadcastCatch audioBroadcastCatch = audioBroadcastManager.get(sendRtpItem.getDeviceId(), sendRtpItem.getChannelId());
@@ -1275,7 +1241,7 @@ public class PlayServiceImpl implements IPlayService {
                         logger.error("[命令发送失败] 停止语音对讲: {}", e.getMessage());
                     }
                 }
-            }else {
+            } else {
                 // 向上级平台
                 try {
                     commanderForPlatform.streamByeCmd(parentPlatform, callIdHeader.getCallId());
@@ -1285,4 +1251,105 @@ public class PlayServiceImpl implements IPlayService {
             }
         }
     }
+
+    @Override
+    public void talkCmd(Device device, String channelId, MediaServerItem mediaServerItem, String stream, AudioEvent event) {
+        if (device == null || channelId == null) {
+            return;
+        }
+        // TODO 必须多端口模式才支持语音喊话鹤语音对讲
+        logger.info("[语音对讲] device： {}, channel: {}", device.getDeviceId(), channelId);
+        DeviceChannel deviceChannel = storager.queryChannel(device.getDeviceId(), channelId);
+        if (deviceChannel == null) {
+            logger.warn("开启语音对讲的时候未找到通道： {}", channelId);
+            event.call("开启语音对讲的时候未找到通道");
+            return;
+        }
+        // 查询通道使用状态
+        if (audioBroadcastManager.exit(device.getDeviceId(), channelId)) {
+            SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(device.getDeviceId(), channelId, null, null);
+            if (sendRtpItem != null && sendRtpItem.isOnlyAudio()) {
+                // 查询流是否存在，不存在则认为是异常状态
+                MediaServerItem mediaServer = mediaServerService.getOne(sendRtpItem.getMediaServerId());
+                Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServer, sendRtpItem.getApp(), sendRtpItem.getStream());
+                if (streamReady) {
+                    logger.warn("[语音对讲] 正在语音广播，无法开启语音通话： {}", channelId);
+                    event.call("正在语音广播");
+                    return;
+                } else {
+                    stopAudioBroadcast(device.getDeviceId(), channelId);
+                }
+            }
+        }
+
+        SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(device.getDeviceId(), channelId, stream, null);
+        if (sendRtpItem != null) {
+            MediaServerItem mediaServer = mediaServerService.getOne(sendRtpItem.getMediaServerId());
+            Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServer, "rtp", sendRtpItem.getReceiveStream());
+            if (streamReady) {
+                logger.warn("[语音对讲] 进行中： {}", channelId);
+                event.call("语音对讲进行中");
+                return;
+            } else {
+                stopTalk(device, channelId);
+            }
+        }
+
+        talk(mediaServerItem, device, channelId, stream, (MediaServerItem mediaServerItem1, JSONObject response) -> {
+            logger.info("[语音对讲] 收到设备发来的流");
+        }, eventResult -> {
+            logger.warn("[语音对讲] 失败，{}/{}, 错误码 {} {}", device.getDeviceId(), channelId, eventResult.statusCode, eventResult.msg);
+            event.call("失败，错误码 " + eventResult.statusCode + ", " + eventResult.msg);
+        }, () -> {
+            logger.warn("[语音对讲] 失败，{}/{} 超时", device.getDeviceId(), channelId);
+            event.call("失败，超时 ");
+            stopTalk(device, channelId);
+        }, errorMsg -> {
+            logger.warn("[语音对讲] 失败，{}/{} {}", device.getDeviceId(), channelId, errorMsg);
+            event.call(errorMsg);
+            stopTalk(device, channelId);
+        });
+    }
+
+    private void stopTalk(Device device, String channelId) {
+        stopTalk(device, channelId, null);
+    }
+
+    @Override
+    public void stopTalk(Device device, String channelId, Boolean streamIsReady) {
+        logger.info("[语音对讲] 停止， {}/{}", device.getDeviceId(), channelId);
+        SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(device.getDeviceId(), channelId, null, null);
+        if (sendRtpItem == null) {
+            logger.info("[语音对讲] 停止失败， 未找到发送信息，可能已经停止");
+            return;
+        }
+        // 停止向设备推流
+        String mediaServerId = sendRtpItem.getMediaServerId();
+        if (mediaServerId == null) {
+            return;
+        }
+
+        MediaServerItem mediaServer = mediaServerService.getOne(mediaServerId);
+
+        if (streamIsReady == null || streamIsReady) {
+            Map<String, Object> param = new HashMap<>();
+            param.put("vhost", "__defaultVhost__");
+            param.put("app", sendRtpItem.getApp());
+            param.put("stream", sendRtpItem.getStream());
+            param.put("ssrc", sendRtpItem.getSsrc());
+            zlmrtpServerFactory.stopSendRtpStream(mediaServer, param);
+        }
+
+        mediaServer.getSsrcConfig().releaseSsrc(sendRtpItem.getSsrc());
+
+        SsrcTransaction ssrcTransaction = streamSession.getSsrcTransaction(device.getDeviceId(), channelId, null, sendRtpItem.getStream());
+        if (ssrcTransaction != null) {
+            try {
+                cmder.streamByeCmd(device, channelId, sendRtpItem.getStream(), null);
+            } catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException  e) {
+                logger.info("[语音对讲] 停止消息发送失败，可能已经停止");
+            }
+        }
+        redisCatchStorage.deleteSendRTPServer(device.getDeviceId(), channelId,null, null);
+    }
 }

src/main/java/com/genersoft/iot/vmp/storager/impl/RedisCatchStorageImpl.java

@@ -378,7 +378,7 @@ public class RedisCatchStorageImpl implements IRedisCatchStorage {
                 + sendRtpItem.getMediaServerId() + "_"
                 + sendRtpItem.getPlatformId() + "_"
                 + sendRtpItem.getChannelId() + "_"
-                + sendRtpItem.getStreamId() + "_"
+                + sendRtpItem.getStream() + "_"
                 + sendRtpItem.getCallId();
         RedisUtil.set(key, sendRtpItem);
     }

src/main/java/com/genersoft/iot/vmp/vmanager/gb28181/play/PlayController.java

@@ -19,11 +19,7 @@ import com.genersoft.iot.vmp.service.IMediaService;
 import com.genersoft.iot.vmp.service.IPlayService;
 import com.genersoft.iot.vmp.storager.IRedisCatchStorage;
 import com.genersoft.iot.vmp.storager.IVideoManagerStorage;
-import com.genersoft.iot.vmp.vmanager.bean.DeferredResultEx;
+import com.genersoft.iot.vmp.vmanager.bean.*;
-import com.genersoft.iot.vmp.vmanager.bean.AudioBroadcastResult;
-import com.genersoft.iot.vmp.vmanager.bean.ErrorCode;
-import com.genersoft.iot.vmp.vmanager.bean.StreamContent;
-import com.genersoft.iot.vmp.vmanager.bean.WVPResult;
 import io.swagger.v3.oas.annotations.Operation;
 import io.swagger.v3.oas.annotations.Parameter;
 import io.swagger.v3.oas.annotations.tags.Tag;
@@ -253,7 +249,7 @@ public class PlayController {
 	@Parameter(name = "timeout", description = "推流超时时间(秒)", required = true)
 	@GetMapping("/broadcast/{deviceId}/{channelId}")
 	@PostMapping("/broadcast/{deviceId}/{channelId}")
-    public AudioBroadcastResult broadcastApi(@PathVariable String deviceId, @PathVariable String channelId, Integer timeout) {
+    public AudioBroadcastResult broadcastApi(@PathVariable String deviceId, @PathVariable String channelId, Integer timeout, Boolean broadcastMode) {
         if (logger.isDebugEnabled()) {
             logger.debug("语音广播API调用");
         }
@@ -265,15 +261,7 @@ public class PlayController {
 			throw new ControllerException(ErrorCode.ERROR400.getCode(), "未找到通道： " + channelId);
 		}
 
-		return playService.audioBroadcast(device, channelId);
+		return playService.audioBroadcast(device, channelId, broadcastMode);
-
-	}
-
-	@GetMapping("/1111")
-	public void broadcastApi1() {
-		MediaServerItem defaultMediaServer = mediaServerService.getMediaServerForMinimumLoad(null);
-		Device device = storager.queryVideoDevice("34020000001320090001");
-		playService.talk(defaultMediaServer, device, "34020000001370000001", null, null, null);
 
 	}
 
@@ -289,7 +277,7 @@ public class PlayController {
 		}
 //		try {
 //			playService.stopAudioBroadcast(deviceId, channelId);
-//		} catch (InvalidArgumentException | ParseException | SsrcTransactionNotFoundException | SipException e) {
+//		} catch (InvalidArgumentException | ParseException  | SipException e) {
 //			logger.error("[命令发送失败] 停止语音: {}", e.getMessage());
 //			throw new ControllerException(ErrorCode.ERROR100.getCode(), "命令发送失败: " +  e.getMessage());
 //		}

src/main/java/com/genersoft/iot/vmp/vmanager/gb28181/play/bean/AudioEvent.java

@@ -4,6 +4,6 @@ package com.genersoft.iot.vmp.vmanager.gb28181.play.bean;
 /**
  * @author lin
  */
-public interface AudioBroadcastEvent {
+public interface AudioEvent {
     void call(String msg);
 }

src/main/java/com/genersoft/iot/vmp/web/gb28181/ApiStreamController.java

@@ -185,7 +185,7 @@ public class ApiStreamController {
         }
         try {
             cmder.streamByeCmd(device, code, streamInfo.getStream(), null);
-        } catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
+        } catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException  e) {
             JSONObject result = new JSONObject();
             result.put("error","发送BYE失败：" + e.getMessage());
             return result;

web_src/config/index.js

@@ -12,7 +12,7 @@ module.exports = {
     assetsPublicPath: './',
     proxyTable: {
       '/debug': {
-        target: 'https://default.wvp-pro.cn:18080',
+        target: 'https://default.wvp-pro.cn:18082',
         changeOrigin: true,
         pathRewrite: {
           '^/debug': '/'

web_src/src/components/dialog/devicePlayer.vue

@@ -299,6 +299,10 @@
 
           </el-tab-pane>
           <el-tab-pane label="语音对讲" name="broadcast">
+            <div style="padding: 0 10px">
+              <el-switch v-model="broadcastMode" :disabled="broadcastStatus !== -1" active-color="#409EFF" active-text="喊话"
+                         inactive-text="对讲"></el-switch>
+            </div>
             <div class="trank" style="text-align: center;">
               <el-button @click="broadcastStatusClick()" :type="getBroadcastStatus()" :disabled="broadcastStatus === -2"
                          circle icon="el-icon-microphone" style="font-size: 32px; padding: 24px;margin-top: 24px;"/>
@@ -390,6 +394,7 @@ export default {
       recordStartTime: 0,
       showTimeText: "00:00:00",
       streamInfo: null,
+      broadcastMode: true,
       broadcastRtc: null,
       broadcastStatus: -1, // -2 正在释放资源 -1 默认状态 0 等待接通 1 接通成功
     };
@@ -648,7 +653,7 @@ export default {
         // 发起语音对讲
         this.$axios({
           method: 'get',
-          url: '/api/play/broadcast/' + this.deviceId + '/' + this.channelId + "?timeout=30"
+          url: '/api/play/broadcast/' + this.deviceId + '/' + this.channelId + "?timeout=30&broadcastMode=" + this.broadcastMode
         }).then( (res)=> {
           if (res.data.code == 0) {
             let streamInfo = res.data.data.streamInfo;