Merge branch 'main' into main2

648540858

2023-01-05 6fa5b37b962b05b3aa4b7bf019eb47c4cdbb5738

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

@@ -1,50 +1,46 @@
package com.genersoft.iot.vmp.gb28181.transmit.event.request.impl;

import com.alibaba.fastjson.JSON;
import com.alibaba.fastjson.JSONObject;
import com.genersoft.iot.vmp.common.StreamInfo;
import com.alibaba.fastjson2.JSONObject;
import com.genersoft.iot.vmp.conf.DynamicTask;
import com.genersoft.iot.vmp.gb28181.bean.InviteStreamType;
import com.genersoft.iot.vmp.conf.UserSetting;
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.cmd.ISIPCommander;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.ISIPCommanderForPlatform;
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.ZLMHttpHookSubscribe;
import com.genersoft.iot.vmp.media.zlm.ZLMRTPServerFactory;
import com.genersoft.iot.vmp.media.zlm.ZlmHttpHookSubscribe;
import com.genersoft.iot.vmp.media.zlm.dto.MediaServerItem;
import com.genersoft.iot.vmp.service.IMediaServerService;
import com.genersoft.iot.vmp.service.IPlayService;
import com.genersoft.iot.vmp.service.bean.RequestPushStreamMsg;
import com.genersoft.iot.vmp.service.impl.RedisGbPlayMsgListener;
import com.genersoft.iot.vmp.service.redisMsg.RedisGbPlayMsgListener;
import com.genersoft.iot.vmp.storager.IRedisCatchStorage;
import com.genersoft.iot.vmp.storager.IVideoManagerStorage;
import com.genersoft.iot.vmp.utils.SerializeUtils;
import org.ehcache.shadow.org.terracotta.offheapstore.storage.IntegerStorageEngine;
import org.slf4j.Logger;
import org.slf4j.LoggerFactory;
import org.springframework.beans.factory.InitializingBean;
import org.springframework.beans.factory.annotation.Autowired;
import org.springframework.stereotype.Component;

import javax.sip.Dialog;
import javax.sip.DialogState;
import javax.sip.RequestEvent;
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.util.*;
import java.text.ParseException;
import java.util.HashMap;
import java.util.Map;

/**
 * SIP命令类型： ACK请求
 * @author lin
 */
@Component
public class AckRequestProcessor extends SIPRequestProcessorParent implements InitializingBean, ISIPRequestProcessor {

   private Logger logger = LoggerFactory.getLogger(AckRequestProcessor.class);
   private final Logger logger = LoggerFactory.getLogger(AckRequestProcessor.class);
   private final String method = "ACK";

   @Autowired
@@ -66,73 +62,61 @@
   private ZLMRTPServerFactory zlmrtpServerFactory;

   @Autowired
   private IMediaServerService mediaServerService;
   private ZlmHttpHookSubscribe hookSubscribe;

   @Autowired
   private ZLMHttpHookSubscribe subscribe;
   private IMediaServerService mediaServerService;

   @Autowired
   private DynamicTask dynamicTask;

   @Autowired
   private ISIPCommander cmder;

   @Autowired
   private ISIPCommanderForPlatform commanderForPlatform;

   @Autowired
   private RedisGbPlayMsgListener redisGbPlayMsgListener;

   @Autowired
   private UserSetting userSetting;

   @Autowired
   private IPlayService playService;


   /**   
    * 处理  ACK请求
    * 
    * @param evt
    */
   @Override
   public void process(RequestEvent evt) {
      Dialog dialog = evt.getDialog();
      CallIdHeader callIdHeader = (CallIdHeader)evt.getRequest().getHeader(CallIdHeader.NAME);
      if (dialog == null) {
         return;
      }
      if (dialog.getState()== DialogState.CONFIRMED) {
         String platformGbId = ((SipURI) ((HeaderAddress) evt.getRequest().getHeader(FromHeader.NAME)).getAddress().getURI()).getUser();
         logger.info("ACK请求： platformGbId->{}", platformGbId);

      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(platformGbId, channelId, null, callIdHeader.getCallId());
         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 is_Udp = sendRtpItem.isTcp() ? "0" : "1";
         MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
         logger.info("收到ACK，开始向上级推流 rtp/{}", sendRtpItem.getStreamId());
         Map<String, Object> param = new HashMap<>();
         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("is_udp", is_Udp);
         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");
         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());
         if (mediaInfo == null) {
            RequestPushStreamMsg requestPushStreamMsg = RequestPushStreamMsg.getInstance(
                  sendRtpItem.getMediaServerId(), sendRtpItem.getApp(), sendRtpItem.getStreamId(),
                  sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isTcp(),
                  sendRtpItem.getLocalPort(), sendRtpItem.getPt(), sendRtpItem.isUsePs(), sendRtpItem.isOnlyAudio());
            redisGbPlayMsgListener.sendMsgForStartSendRtpStream(sendRtpItem.getServerId(), requestPushStreamMsg, jsonObject->{
               startSendRtpStreamHand(evt, sendRtpItem, parentPlatform, jsonObject, param, callIdHeader);
            redisGbPlayMsgListener.sendMsgForStartSendRtpStream(sendRtpItem.getServerId(), requestPushStreamMsg, json -> {
               startSendRtpStreamHand(evt, sendRtpItem, parentPlatform, json, callIdHeader);
            });
         }else {
            JSONObject jsonObject = zlmrtpServerFactory.startSendRtpStream(mediaInfo, param);
            startSendRtpStreamHand(evt, sendRtpItem, parentPlatform, jsonObject, param, callIdHeader);
            JSONObject startSendRtpStreamResult = zlmrtpServerFactory.startSendRtp(mediaInfo, sendRtpItem);
            if (startSendRtpStreamResult != null) {
               startSendRtpStreamHand(evt, sendRtpItem, parentPlatform, startSendRtpStreamResult, callIdHeader);
            }
         }


      }
   }
   private void startSendRtpStreamHand(RequestEvent evt, SendRtpItem sendRtpItem, ParentPlatform parentPlatform,
@@ -140,20 +124,24 @@
      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("RTP推流成功[ {}/{} ]，{}->{}:{}, " ,param.get("app"), param.get("stream"), jsonObject.getString("local_port"), param.get("dst_url"), param.get("dst_port"));
         byte[] dialogByteArray = SerializeUtils.serialize(evt.getDialog());
         sendRtpItem.setDialog(dialogByteArray);
         byte[] transactionByteArray = SerializeUtils.serialize(evt.getServerTransaction());
         sendRtpItem.setTransaction(transactionByteArray);
         redisCatchStorage.updateSendRTPSever(sendRtpItem);
      } else {
         logger.error("RTP推流失败: {}, 参数：{}",jsonObject.getString("msg"),JSONObject.toJSON(param));
         logger.error("RTP推流失败: {}, 参数：{}",jsonObject.getString("msg"), JSON.toJSONString(param));
         if (sendRtpItem.isOnlyAudio()) {
            // TODO 可能是语音对讲
         }else {
            // 向上级平台
            commanderForPlatform.streamByeCmd(parentPlatform, callIdHeader.getCallId());
            Device device = deviceService.getDevice(sendRtpItem.getDeviceId());
            AudioBroadcastCatch audioBroadcastCatch = audioBroadcastManager.get(sendRtpItem.getDeviceId(), sendRtpItem.getChannelId());
            if (audioBroadcastCatch != null) {
               try {
                  cmder.streamByeCmd(device, sendRtpItem.getChannelId(), audioBroadcastCatch.getSipTransactionInfo(), null);
               } catch (SipException | ParseException | InvalidArgumentException |
                      SsrcTransactionNotFoundException e) {
                  logger.error("[命令发送失败] 停止语音对讲: {}", e.getMessage());
               }
            }
         }
      }

   }

}