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648540858

2023-01-05 f275daa3f86f0cbcbe3176e7942994c3a9869480

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

@@ -10,6 +10,8 @@
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.service.IPlayService;
@@ -29,7 +31,6 @@
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;

@@ -99,49 +100,62 @@
            logger.warn("[收到ACK]：未找到通道({})的推流信息", channelId);
            return;
         }
         String is_Udp = sendRtpItem.isTcp() ? "0" : "1";
         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) {
            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, json -> {
               startSendRtpStreamHand(evt, sendRtpItem, parentPlatform, json, callIdHeader);
               playService.startSendRtpStreamHand(sendRtpItem, parentPlatform, json, param, callIdHeader);
            });
         }else {
            JSONObject startSendRtpStreamResult = zlmrtpServerFactory.startSendRtp(mediaInfo, sendRtpItem);
            if (startSendRtpStreamResult != null) {
               startSendRtpStreamHand(evt, sendRtpItem, parentPlatform, startSendRtpStreamResult, callIdHeader);
            }
         }
      }
   }
   private void startSendRtpStreamHand(RequestEvent evt, SendRtpItem sendRtpItem, ParentPlatform parentPlatform,
                              JSONObject jsonObject, Map<String, Object> param, CallIdHeader callIdHeader) {
      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"));
      } else {
         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());
            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());
         } 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 {
               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);
               }
            }
            if (startSendRtpStreamResult != null) {
               playService.startSendRtpStreamHand(sendRtpItem, parentPlatform, startSendRtpStreamResult, param, callIdHeader);
            }
         }
      }

   }

}