添加第二种语音对讲实现

648540858

2022-09-29 06bbe3fe01e5af9486c309693a975077df813f7c

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

@@ -11,12 +11,16 @@
import com.genersoft.iot.vmp.conf.exception.SsrcTransactionNotFoundException;

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.impl.SIPCommanderFroPlatform;

import com.genersoft.iot.vmp.gb28181.transmit.callback.RequestMessage;

import com.genersoft.iot.vmp.gb28181.transmit.cmd.ISIPCommanderForPlatform;

import com.genersoft.iot.vmp.media.zlm.dto.*;

import com.genersoft.iot.vmp.service.*;

import com.genersoft.iot.vmp.storager.IRedisCatchStorage;

import com.genersoft.iot.vmp.storager.IVideoManagerStorage;

import com.genersoft.iot.vmp.vmanager.bean.AudioBroadcastResult;

import com.genersoft.iot.vmp.vmanager.bean.WVPResult;

import org.slf4j.Logger;

import org.slf4j.LoggerFactory;

import org.springframework.beans.factory.annotation.Autowired;

@@ -51,7 +55,13 @@
   private SIPCommander cmder;



   @Autowired

   private SIPCommanderFroPlatform commanderFroPlatform;

   private ISIPCommanderForPlatform commanderFroPlatform;



   @Autowired

   private AudioBroadcastManager audioBroadcastManager;



   @Autowired

   private ZLMRTPServerFactory zlmrtpServerFactory;



   @Autowired

   private IPlayService playService;

@@ -466,7 +476,127 @@
                        streamInfo.getStream(), null);

               }

            }

         }else {

         }else if ("broadcast".equals(app)){

            // 语音对讲推流  stream需要满足格式deviceId_channelId

            if (regist && stream.indexOf("_") > 0) {

               String[] streamArray = stream.split("_");

               if (streamArray.length == 2) {

                  String deviceId = streamArray[0];

                  String channelId = streamArray[1];

                  Device device = deviceService.queryDevice(deviceId);

                  if (device != null) {

                     DeviceChannel deviceChannel = storager.queryChannel(deviceId, channelId);

                     if (deviceChannel != null) {

                        if (audioBroadcastManager.exit(deviceId, channelId)) {

                           // 直接推流

                           SendRtpItem sendRtpItem =  redisCatchStorage.querySendRTPServer(null, null, stream, null);

                           if (sendRtpItem == null) {

                              // TODO 可能数据错误，重新开启语音通道

                           }else {

                              String is_Udp = sendRtpItem.isTcp() ? "0" : "1";

                              MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());

                              logger.info("rtp/{}开始向上级推流, 目标={}:{}，SSRC={}", sendRtpItem.getStreamId(), sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc());

                              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("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");



                              JSONObject jsonObject;

                              if (sendRtpItem.isTcpActive()) {

                                 jsonObject = zlmrtpServerFactory.startSendRtpPassive(mediaInfo, param);

                              } else {

                                 param.put("is_udp", is_Udp);

                                 param.put("dst_url", sendRtpItem.getIp());

                                 param.put("dst_port", sendRtpItem.getPort());

                                 jsonObject = zlmrtpServerFactory.startSendRtpStream(mediaInfo, param);

                              }

                              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

                           try {

                              playService.audioBroadcastCmd(device, channelId, 60, (msg)->{

                                 logger.info("[语音对讲] 通道建立成功, device: {}, channel: {}", deviceId, channelId);

                              });

                           } catch (InvalidArgumentException | ParseException | SipException e) {

                              logger.error("[命令发送失败] 语音对讲: {}", e.getMessage());

                           }

                        }



                     }

                  }

               }

            }



         }else if ("talk".equals(app)){

            // 语音对讲推流  stream需要满足格式deviceId_channelId

            if (regist && stream.indexOf("_") > 0) {

               String[] streamArray = stream.split("_");

               if (streamArray.length == 2) {

                  String deviceId = streamArray[0];

                  String channelId = streamArray[1];

                  Device device = deviceService.queryDevice(deviceId);

                  if (device != null) {

                     DeviceChannel deviceChannel = storager.queryChannel(deviceId, channelId);

                     if (deviceChannel != null) {

                        if (audioBroadcastManager.exit(deviceId, channelId)) {

                           // 直接推流

                           SendRtpItem sendRtpItem =  redisCatchStorage.querySendRTPServer(null, null, stream, null);

                           if (sendRtpItem == null) {

                              // TODO 可能数据错误，重新开启语音通道

                           }else {

                              String is_Udp = sendRtpItem.isTcp() ? "0" : "1";

                              MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());

                              logger.info("rtp/{}开始向上级推流, 目标={}:{}，SSRC={}", sendRtpItem.getStreamId(), sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc());

                              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("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");



                              JSONObject jsonObject;

                              if (sendRtpItem.isTcpActive()) {

                                 jsonObject = zlmrtpServerFactory.startSendRtpPassive(mediaInfo, param);

                              } else {

                                 param.put("is_udp", is_Udp);

                                 param.put("dst_url", sendRtpItem.getIp());

                                 param.put("dst_port", sendRtpItem.getPort());

                                 jsonObject = zlmrtpServerFactory.startSendRtpStream(mediaInfo, param);

                              }

                              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 mediaServerForMinimumLoad = mediaServerService.getMediaServerForMinimumLoad();

                           playService.talk(mediaServerForMinimumLoad, device, channelId, (mediaServerItem, jsonObject)->{

                              System.out.println("开始推流");

                           }, eventResult -> {

                              System.out.println(eventResult.msg);

                           }, ()->{

                              System.out.println("超时");

                           });

                        }



                     }

                  }

               }

            }



         }else{

            if (!"rtp".equals(app)){

               String type = OriginType.values()[item.getOriginType()].getType();

               MediaServerItem mediaServerItem = mediaServerService.getOne(mediaServerId);

@@ -521,10 +651,23 @@
                  if (sendRtpItem.getApp().equals(app)) {

                     String platformId = sendRtpItem.getPlatformId();

                     ParentPlatform platform = storager.queryParentPlatByServerGBId(platformId);

                     Device device = deviceService.queryDevice(platformId);



                     try {

                        commanderFroPlatform.streamByeCmd(platform, sendRtpItem);

                     } catch (SipException | InvalidArgumentException | ParseException e) {

                        if (platform != null) {

                           commanderFroPlatform.streamByeCmd(platform, sendRtpItem);

                        }else {

                           if (sendRtpItem.isOnlyAudio()) {

                              AudioBroadcastCatch audioBroadcastCatch = audioBroadcastManager.get(sendRtpItem.getDeviceId(), sendRtpItem.getChannelId());

                              if (device != null && audioBroadcastCatch != null) {

//                                 cmder.streamByeCmd(device, sendRtpItem.getChannelId(), audioBroadcastCatch.getSipTransactionInfo(), null);

                              }

                           }else {

                              cmder.streamByeCmd(device, sendRtpItem.getChannelId(), stream, sendRtpItem.getCallId());

                           }



                        }

                     } catch (SipException | InvalidArgumentException | ParseException | SsrcTransactionNotFoundException e) {

                        logger.error("[命令发送失败] 国标级联 发送BYE: {}", e.getMessage());

                     }

                  }