Merge branch 'refs/heads/master' into develop-add-api-key

leesam

2024-04-10 16b7e4a7ef473a6af29ec78aeb2f471fa398efdd

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

@@ -3,17 +3,20 @@
import com.alibaba.fastjson2.JSONObject;
import com.genersoft.iot.vmp.conf.DynamicTask;
import com.genersoft.iot.vmp.conf.UserSetting;
import com.genersoft.iot.vmp.conf.exception.ControllerException;
import com.genersoft.iot.vmp.conf.exception.SsrcTransactionNotFoundException;
import com.genersoft.iot.vmp.gb28181.bean.AudioBroadcastCatch;
import com.genersoft.iot.vmp.gb28181.bean.Device;
import com.genersoft.iot.vmp.gb28181.bean.ParentPlatform;
import com.genersoft.iot.vmp.gb28181.bean.SendRtpItem;
import com.genersoft.iot.vmp.gb28181.transmit.SIPProcessorObserver;
import com.genersoft.iot.vmp.gb28181.transmit.event.request.ISIPRequestProcessor;
import com.genersoft.iot.vmp.gb28181.transmit.event.request.SIPRequestProcessorParent;
import com.genersoft.iot.vmp.media.zlm.ZLMRTPServerFactory;
import com.genersoft.iot.vmp.media.zlm.ZlmHttpHookSubscribe;
import com.genersoft.iot.vmp.media.zlm.dto.HookSubscribeFactory;
import com.genersoft.iot.vmp.media.zlm.dto.HookSubscribeForRtpServerTimeout;
import com.genersoft.iot.vmp.media.zlm.dto.MediaServerItem;
import com.genersoft.iot.vmp.service.IMediaServerService;
import com.genersoft.iot.vmp.media.zlm.ZLMServerFactory;
import com.genersoft.iot.vmp.media.event.hook.HookSubscribe;
import com.genersoft.iot.vmp.media.bean.MediaServer;
import com.genersoft.iot.vmp.service.IDeviceService;
import com.genersoft.iot.vmp.media.service.IMediaServerService;
import com.genersoft.iot.vmp.service.IPlayService;
import com.genersoft.iot.vmp.service.bean.RequestPushStreamMsg;
import com.genersoft.iot.vmp.service.redisMsg.RedisGbPlayMsgListener;
@@ -25,12 +28,15 @@
import org.springframework.beans.factory.annotation.Autowired;
import org.springframework.stereotype.Component;

import javax.sip.InvalidArgumentException;
import javax.sip.RequestEvent;
import javax.sip.SipException;
import javax.sip.address.SipURI;
import javax.sip.header.CallIdHeader;
import javax.sip.header.FromHeader;
import javax.sip.header.HeaderAddress;
import javax.sip.header.ToHeader;
import java.text.ParseException;
import java.util.HashMap;
import java.util.Map;

@@ -57,13 +63,19 @@
    private IRedisCatchStorage redisCatchStorage;

   @Autowired
    private UserSetting userSetting;

   @Autowired
   private IVideoManagerStorage storager;

   @Autowired
   private ZLMRTPServerFactory zlmrtpServerFactory;
   private IDeviceService deviceService;

   @Autowired
   private ZlmHttpHookSubscribe hookSubscribe;
   private ZLMServerFactory zlmServerFactory;

   @Autowired
   private HookSubscribe hookSubscribe;

   @Autowired
   private IMediaServerService mediaServerService;
@@ -75,9 +87,6 @@
   private RedisGbPlayMsgListener redisGbPlayMsgListener;

   @Autowired
   private UserSetting userSetting;

   @Autowired
   private IPlayService playService;


@@ -87,76 +96,76 @@
   @Override
   public void process(RequestEvent evt) {
      CallIdHeader callIdHeader = (CallIdHeader)evt.getRequest().getHeader(CallIdHeader.NAME);
      dynamicTask.stop(callIdHeader.getCallId());
      String fromUserId = ((SipURI) ((HeaderAddress) evt.getRequest().getHeader(FromHeader.NAME)).getAddress().getURI()).getUser();
      String toUserId = ((SipURI) ((HeaderAddress) evt.getRequest().getHeader(ToHeader.NAME)).getAddress().getURI()).getUser();
      logger.info("[收到ACK]： 来自->{}", fromUserId);
      SendRtpItem sendRtpItem =  redisCatchStorage.querySendRTPServer(null, null, null, callIdHeader.getCallId());
      if (sendRtpItem == null) {
         logger.warn("[收到ACK]：未找到来自{}，目标为({})的推流信息",fromUserId, toUserId);
         return;
      }
      // tcp主动时，此时是级联下级平台，在回复200ok时，本地已经请求zlm开启监听，跳过下面步骤
      if (sendRtpItem.isTcpActive()) {
         logger.info("收到ACK，rtp/{} TCP主动方式后续处理", sendRtpItem.getStream());
         return;
      }
      MediaServer mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
      logger.info("收到ACK，rtp/{}开始向上级推流, 目标={}:{}，SSRC={}, 协议:{}",
            sendRtpItem.getStream(),
            sendRtpItem.getIp(),
            sendRtpItem.getPort(),
            sendRtpItem.getSsrc(),
            sendRtpItem.isTcp()?(sendRtpItem.isTcpActive()?"TCP主动":"TCP被动"):"UDP"
      );
      ParentPlatform parentPlatform = storager.queryParentPlatByServerGBId(fromUserId);

      String platformGbId = ((SipURI) ((HeaderAddress) evt.getRequest().getHeader(FromHeader.NAME)).getAddress().getURI()).getUser();
      logger.info("[收到ACK]： platformGbId->{}", platformGbId);
      if (userSetting.getPushStreamAfterAck()) {
         ParentPlatform parentPlatform = storager.queryParentPlatByServerGBId(platformGbId);
         // 取消设置的超时任务
         dynamicTask.stop(callIdHeader.getCallId());
         String channelId = ((SipURI) ((HeaderAddress) evt.getRequest().getHeader(ToHeader.NAME)).getAddress().getURI()).getUser();
         SendRtpItem sendRtpItem =  redisCatchStorage.querySendRTPServer(null, null, null, callIdHeader.getCallId());
         if (sendRtpItem == null) {
            logger.warn("[收到ACK]：未找到通道({})的推流信息", channelId);
            return;
         }
         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(),
               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("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", isUdp);
         if (!sendRtpItem.isTcp()) {
            // udp模式下开启rtcp保活
            param.put("udp_rtcp_timeout", sendRtpItem.isRtcp()? "1":"0");
         }

         if (mediaInfo == null) {
      if (parentPlatform != null) {
         if (!userSetting.getServerId().equals(sendRtpItem.getServerId())) {
            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 -> {
               playService.startSendRtpStreamHand(sendRtpItem, parentPlatform, json, param, callIdHeader);
               playService.startSendRtpStreamFailHand(sendRtpItem, parentPlatform, callIdHeader);
            });
         } else {
            // 如果是非严格模式，需要关闭端口占用
            JSONObject startSendRtpStreamResult = null;
            if (sendRtpItem.getLocalPort() != 0) {
               HookSubscribeForRtpServerTimeout hookSubscribeForRtpServerTimeout = HookSubscribeFactory.on_rtp_server_timeout(sendRtpItem.getSsrc(), null, mediaInfo.getId());
               hookSubscribe.removeSubscribe(hookSubscribeForRtpServerTimeout);
               if (zlmrtpServerFactory.releasePort(mediaInfo, sendRtpItem.getSsrc())) {
                  if (sendRtpItem.isTcpActive()) {
                     startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpPassive(mediaInfo, param);
                  }else {
                     param.put("dst_url", sendRtpItem.getIp());
                     param.put("dst_port", sendRtpItem.getPort());
                     startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpStream(mediaInfo, param);
                  }
               }
            }else {
            try {
               if (sendRtpItem.isTcpActive()) {
                  startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpPassive(mediaInfo, param);
               }else {
                  param.put("dst_url", sendRtpItem.getIp());
                  param.put("dst_port", sendRtpItem.getPort());
                  startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpStream(mediaInfo, param);
                  mediaServerService.startSendRtpPassive(mediaInfo, parentPlatform, sendRtpItem, null);
               } else {
                  mediaServerService.startSendRtpStream(mediaInfo, parentPlatform, sendRtpItem);
               }
            }
            if (startSendRtpStreamResult != null) {
               playService.startSendRtpStreamHand(sendRtpItem, parentPlatform, startSendRtpStreamResult, param, callIdHeader);
            }catch (ControllerException e) {
               logger.error("RTP推流失败: {}", e.getMessage());
               playService.startSendRtpStreamFailHand(sendRtpItem, parentPlatform, callIdHeader);
            }
         }
      }else {
         Device device = deviceService.getDevice(fromUserId);
         if (device == null) {
            logger.warn("[收到ACK]：来自{}，目标为({})的推流信息为找到流体服务[{}]信息",fromUserId, toUserId, sendRtpItem.getMediaServerId());
            return;
         }
         // 设置为收到ACK后发送语音的设备已经在发送200OK开始发流了
         if (!device.isBroadcastPushAfterAck()) {
            return;
         }
         if (mediaInfo == null) {
            logger.warn("[收到ACK]：来自{}，目标为({})的推流信息为找到流体服务[{}]信息",fromUserId, toUserId, sendRtpItem.getMediaServerId());
            return;
         }
         try {
            if (sendRtpItem.isTcpActive()) {
               mediaServerService.startSendRtpPassive(mediaInfo, null, sendRtpItem, null);
            } else {
               mediaServerService.startSendRtpStream(mediaInfo, null, sendRtpItem);
            }
         }catch (ControllerException e) {
            logger.error("RTP推流失败: {}", e.getMessage());
            playService.startSendRtpStreamFailHand(sendRtpItem, null, callIdHeader);
         }
      }

   }

}