合并主线

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.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.service.IPlayService;
@@ -29,7 +31,6 @@ 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;
 
@@ -99,49 +100,62 @@ public class AckRequestProcessor extends SIPRequestProcessorParent implements In
 				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));
+				// 如果是非严格模式，需要关闭端口占用
-			if (sendRtpItem.isOnlyAudio()) {
+				JSONObject startSendRtpStreamResult = null;
-				Device device = deviceService.getDevice(sendRtpItem.getDeviceId());
+				if (sendRtpItem.getLocalPort() != 0) {
-				AudioBroadcastCatch audioBroadcastCatch = audioBroadcastManager.get(sendRtpItem.getDeviceId(), sendRtpItem.getChannelId());
+					HookSubscribeForRtpServerTimeout hookSubscribeForRtpServerTimeout = HookSubscribeFactory.on_rtp_server_timeout(sendRtpItem.getSsrc(), null, mediaInfo.getId());
-				if (audioBroadcastCatch != null) {
+					hookSubscribe.removeSubscribe(hookSubscribeForRtpServerTimeout);
-					try {
+					if (zlmrtpServerFactory.releasePort(mediaInfo, sendRtpItem.getSsrc())) {
-						cmder.streamByeCmd(device, sendRtpItem.getChannelId(), audioBroadcastCatch.getSipTransactionInfo(), null);
+						if (sendRtpItem.isTcpActive()) {
-					} catch (SipException | ParseException | InvalidArgumentException |
+							startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpPassive(mediaInfo, param);
-							 SsrcTransactionNotFoundException e) {
+						}else {
-						logger.error("[命令发送失败] 停止语音对讲: {}", e.getMessage());
+							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);
 				}
 			}
 		}
 	}
 
-	}
-
 }

src/main/java/com/genersoft/iot/vmp/service/impl/PlayServiceImpl.java

@@ -1035,12 +1035,12 @@ public class PlayServiceImpl implements IPlayService {
             SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(device.getDeviceId(), channelId, null, null);
             if (sendRtpItem != null && sendRtpItem.isOnlyAudio()) {
                 // 查询流是否存在，不存在则认为是异常状态
-                MediaServerItem mediaServerItem = mediaServerService.getOne(sendRtpItem.getMediaServerId());
+                MediaServerItem mediaServerItemInUse = mediaServerService.getOne(sendRtpItem.getMediaServerId());
-                Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServerItem, sendRtpItem.getApp(), sendRtpItem.getStreamId());
+                Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServerItemInUse, sendRtpItem.getApp(), sendRtpItem.getStreamId());
                 if (streamReady) {
                     logger.warn("语音广播已经开启： {}", channelId);
                     event.call("语音广播已经开启");
-                    return;
+                    return false;
                 } else {
                     stopAudioBroadcast(device.getDeviceId(), channelId);
                 }