优化语音广播的TCP主动模式

648540858

2022-05-18 9c555b56b70ed05c1b13f0a26098b702d7364839

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

@@ -4,9 +4,8 @@
import com.alibaba.fastjson.JSONObject;
import com.genersoft.iot.vmp.common.StreamInfo;
import com.genersoft.iot.vmp.conf.DynamicTask;
import com.genersoft.iot.vmp.gb28181.bean.InviteStreamType;
import com.genersoft.iot.vmp.gb28181.bean.ParentPlatform;
import com.genersoft.iot.vmp.gb28181.bean.SendRtpItem;
import com.genersoft.iot.vmp.gb28181.bean.*;
import com.genersoft.iot.vmp.gb28181.session.AudioBroadcastManager;
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;
@@ -82,6 +81,9 @@
   @Autowired
   private ISIPCommanderForPlatform commanderForPlatform;

   @Autowired
   private AudioBroadcastManager audioBroadcastManager;


   /**   
    * 处理  ACK请求
@@ -105,23 +107,42 @@
         SendRtpItem sendRtpItem =  redisCatchStorage.querySendRTPServer(platformGbId, null, null, callIdHeader.getCallId());
         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());
         logger.info("[收到ACK]，开始使用{}向上级推流 {}/{}->{}:{}({})", sendRtpItem.isTcp() ? "TCP" : "UDP",
               sendRtpItem.getApp(), sendRtpItem.getStreamId(),
               sendRtpItem.getIp() ,sendRtpItem.getPort(),
               sendRtpItem.getSsrc());
         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");
         JSONObject jsonObject = zlmrtpServerFactory.startSendRtpStream(mediaInfo, param);
         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) {
            logger.error("RTP推流失败: 请检查ZLM服务");
         } else if (jsonObject.getInteger("code") == 0) {

            if (sendRtpItem.isOnlyAudio()) {
               AudioBroadcastCatch audioBroadcastCatch = audioBroadcastManager.get(sendRtpItem.getDeviceId(), sendRtpItem.getChannelId());
               audioBroadcastCatch.setStatus(AudioBroadcastCatchStatus.Ok);
               audioBroadcastCatch.setDialog((SIPDialog) evt.getDialog());
               audioBroadcastCatch.setRequest((SIPRequest) evt.getRequest());
               audioBroadcastManager.update(audioBroadcastCatch);
               String waiteStreamTimeoutTaskKey = "waite-stream-" + audioBroadcastCatch.getDeviceId() + audioBroadcastCatch.getChannelId();
               dynamicTask.stop(waiteStreamTimeoutTaskKey);
            }
            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"),JSONObject.toJSON(param));