合并主线
parent
6fa5b37b96
commit
f275daa3f8
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@ -10,6 +10,8 @@ import com.genersoft.iot.vmp.gb28181.transmit.event.request.ISIPRequestProcessor
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import com.genersoft.iot.vmp.gb28181.transmit.event.request.SIPRequestProcessorParent;
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import com.genersoft.iot.vmp.media.zlm.ZLMRTPServerFactory;
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import com.genersoft.iot.vmp.media.zlm.ZlmHttpHookSubscribe;
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import com.genersoft.iot.vmp.media.zlm.dto.HookSubscribeFactory;
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import com.genersoft.iot.vmp.media.zlm.dto.HookSubscribeForRtpServerTimeout;
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import com.genersoft.iot.vmp.media.zlm.dto.MediaServerItem;
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import com.genersoft.iot.vmp.service.IMediaServerService;
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import com.genersoft.iot.vmp.service.IPlayService;
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@ -29,7 +31,6 @@ import javax.sip.header.CallIdHeader;
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import javax.sip.header.FromHeader;
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import javax.sip.header.HeaderAddress;
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import javax.sip.header.ToHeader;
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import java.text.ParseException;
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import java.util.HashMap;
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import java.util.Map;
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@ -99,49 +100,62 @@ public class AckRequestProcessor extends SIPRequestProcessorParent implements In
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logger.warn("[收到ACK]:未找到通道({})的推流信息", channelId);
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return;
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}
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String is_Udp = sendRtpItem.isTcp() ? "0" : "1";
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String isUdp = sendRtpItem.isTcp() ? "0" : "1";
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MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
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logger.info("收到ACK,rtp/{}开始向上级推流, 目标={}:{},SSRC={}, RTCP={}", sendRtpItem.getStreamId(),
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sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isRtcp());
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Map<String, Object> param = new HashMap<>(12);
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param.put("vhost","__defaultVhost__");
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param.put("app",sendRtpItem.getApp());
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param.put("stream",sendRtpItem.getStreamId());
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param.put("ssrc", sendRtpItem.getSsrc());
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param.put("src_port", sendRtpItem.getLocalPort());
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param.put("pt", sendRtpItem.getPt());
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param.put("use_ps", sendRtpItem.isUsePs() ? "1" : "0");
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param.put("only_audio", sendRtpItem.isOnlyAudio() ? "1" : "0");
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param.put("is_udp", isUdp);
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if (!sendRtpItem.isTcp()) {
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// udp模式下开启rtcp保活
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param.put("udp_rtcp_timeout", sendRtpItem.isRtcp()? "1":"0");
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}
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if (mediaInfo == null) {
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RequestPushStreamMsg requestPushStreamMsg = RequestPushStreamMsg.getInstance(
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sendRtpItem.getMediaServerId(), sendRtpItem.getApp(), sendRtpItem.getStreamId(),
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sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isTcp(),
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sendRtpItem.getLocalPort(), sendRtpItem.getPt(), sendRtpItem.isUsePs(), sendRtpItem.isOnlyAudio());
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redisGbPlayMsgListener.sendMsgForStartSendRtpStream(sendRtpItem.getServerId(), requestPushStreamMsg, json -> {
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startSendRtpStreamHand(evt, sendRtpItem, parentPlatform, json, callIdHeader);
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playService.startSendRtpStreamHand(sendRtpItem, parentPlatform, json, param, callIdHeader);
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});
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}else {
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JSONObject startSendRtpStreamResult = zlmrtpServerFactory.startSendRtp(mediaInfo, sendRtpItem);
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if (startSendRtpStreamResult != null) {
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startSendRtpStreamHand(evt, sendRtpItem, parentPlatform, startSendRtpStreamResult, callIdHeader);
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}
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}
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}
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}
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private void startSendRtpStreamHand(RequestEvent evt, SendRtpItem sendRtpItem, ParentPlatform parentPlatform,
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JSONObject jsonObject, Map<String, Object> param, CallIdHeader callIdHeader) {
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if (jsonObject == null) {
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logger.error("RTP推流失败: 请检查ZLM服务");
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} else if (jsonObject.getInteger("code") == 0) {
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logger.info("调用ZLM推流接口, 结果: {}", jsonObject);
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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"));
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} else {
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logger.error("RTP推流失败: {}, 参数:{}",jsonObject.getString("msg"), JSON.toJSONString(param));
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if (sendRtpItem.isOnlyAudio()) {
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Device device = deviceService.getDevice(sendRtpItem.getDeviceId());
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AudioBroadcastCatch audioBroadcastCatch = audioBroadcastManager.get(sendRtpItem.getDeviceId(), sendRtpItem.getChannelId());
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if (audioBroadcastCatch != null) {
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try {
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cmder.streamByeCmd(device, sendRtpItem.getChannelId(), audioBroadcastCatch.getSipTransactionInfo(), null);
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} catch (SipException | ParseException | InvalidArgumentException |
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SsrcTransactionNotFoundException e) {
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logger.error("[命令发送失败] 停止语音对讲: {}", e.getMessage());
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} else {
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// 如果是非严格模式,需要关闭端口占用
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JSONObject startSendRtpStreamResult = null;
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if (sendRtpItem.getLocalPort() != 0) {
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HookSubscribeForRtpServerTimeout hookSubscribeForRtpServerTimeout = HookSubscribeFactory.on_rtp_server_timeout(sendRtpItem.getSsrc(), null, mediaInfo.getId());
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hookSubscribe.removeSubscribe(hookSubscribeForRtpServerTimeout);
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if (zlmrtpServerFactory.releasePort(mediaInfo, sendRtpItem.getSsrc())) {
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if (sendRtpItem.isTcpActive()) {
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startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpPassive(mediaInfo, param);
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}else {
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param.put("dst_url", sendRtpItem.getIp());
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param.put("dst_port", sendRtpItem.getPort());
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startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpStream(mediaInfo, param);
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}
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}
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}else {
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if (sendRtpItem.isTcpActive()) {
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startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpPassive(mediaInfo, param);
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}else {
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param.put("dst_url", sendRtpItem.getIp());
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param.put("dst_port", sendRtpItem.getPort());
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startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpStream(mediaInfo, param);
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}
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}
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if (startSendRtpStreamResult != null) {
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playService.startSendRtpStreamHand(sendRtpItem, parentPlatform, startSendRtpStreamResult, param, callIdHeader);
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}
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}
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}
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}
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}
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@ -1035,12 +1035,12 @@ public class PlayServiceImpl implements IPlayService {
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SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(device.getDeviceId(), channelId, null, null);
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if (sendRtpItem != null && sendRtpItem.isOnlyAudio()) {
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// 查询流是否存在,不存在则认为是异常状态
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MediaServerItem mediaServerItem = mediaServerService.getOne(sendRtpItem.getMediaServerId());
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Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServerItem, sendRtpItem.getApp(), sendRtpItem.getStreamId());
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MediaServerItem mediaServerItemInUse = mediaServerService.getOne(sendRtpItem.getMediaServerId());
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Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServerItemInUse, sendRtpItem.getApp(), sendRtpItem.getStreamId());
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if (streamReady) {
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logger.warn("语音广播已经开启: {}", channelId);
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event.call("语音广播已经开启");
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return;
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return false;
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} else {
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stopAudioBroadcast(device.getDeviceId(), channelId);
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}
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