完善语音对讲级联

648540858

2023-03-21 82adc0cb23f3ee47322e78889cdaba57e9309000

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

@@ -5,51 +5,60 @@
import com.alibaba.fastjson2.JSONObject;
import com.genersoft.iot.vmp.common.StreamInfo;
import com.genersoft.iot.vmp.conf.DynamicTask;
import com.genersoft.iot.vmp.conf.SipConfig;
import com.genersoft.iot.vmp.conf.UserSetting;
import com.genersoft.iot.vmp.conf.exception.ControllerException;
import com.genersoft.iot.vmp.conf.exception.ServiceException;
import com.genersoft.iot.vmp.conf.exception.SsrcTransactionNotFoundException;
import com.genersoft.iot.vmp.gb28181.bean.*;
import com.genersoft.iot.vmp.gb28181.event.SipSubscribe;
import com.genersoft.iot.vmp.gb28181.session.AudioBroadcastManager;
import com.genersoft.iot.vmp.gb28181.session.VideoStreamSessionManager;
import com.genersoft.iot.vmp.gb28181.transmit.callback.DeferredResultHolder;
import com.genersoft.iot.vmp.gb28181.transmit.callback.RequestMessage;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.ISIPCommanderForPlatform;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.impl.SIPCommander;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.impl.SIPCommanderFroPlatform;
import com.genersoft.iot.vmp.gb28181.utils.SipUtils;
import com.genersoft.iot.vmp.media.zlm.AssistRESTfulUtils;
import com.genersoft.iot.vmp.media.zlm.ZLMRESTfulUtils;
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.HookSubscribeForStreamChange;
import com.genersoft.iot.vmp.media.zlm.dto.MediaServerItem;
import com.genersoft.iot.vmp.service.IDeviceService;
import com.genersoft.iot.vmp.service.IMediaServerService;
import com.genersoft.iot.vmp.service.IMediaService;
import com.genersoft.iot.vmp.service.IPlayService;
import com.genersoft.iot.vmp.service.bean.InviteTimeOutCallback;
import com.genersoft.iot.vmp.service.bean.PlayBackCallback;
import com.genersoft.iot.vmp.service.bean.PlayBackResult;
import com.genersoft.iot.vmp.service.bean.SSRCInfo;
import com.genersoft.iot.vmp.service.bean.*;
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.DateUtil;
import com.genersoft.iot.vmp.utils.redis.RedisUtil;
import com.genersoft.iot.vmp.vmanager.bean.AudioBroadcastResult;
import com.genersoft.iot.vmp.vmanager.bean.ErrorCode;
import com.genersoft.iot.vmp.vmanager.bean.StreamContent;
import com.genersoft.iot.vmp.vmanager.bean.WVPResult;
import com.genersoft.iot.vmp.vmanager.gb28181.play.bean.AudioBroadcastEvent;
import gov.nist.javax.sip.message.SIPResponse;
import org.slf4j.Logger;
import org.slf4j.LoggerFactory;
import org.springframework.beans.factory.annotation.Autowired;
import org.springframework.beans.factory.annotation.Qualifier;
import org.springframework.scheduling.concurrent.ThreadPoolTaskExecutor;
import org.springframework.stereotype.Service;
import org.springframework.util.ObjectUtils;

import javax.sip.InvalidArgumentException;
import javax.sip.ResponseEvent;
import javax.sip.SipException;
import javax.sip.header.CallIdHeader;
import java.math.BigDecimal;
import java.math.RoundingMode;
import java.text.ParseException;
import java.util.List;
import java.util.UUID;
import java.util.*;

@SuppressWarnings(value = {"rawtypes", "unchecked"})
@Service
@@ -64,10 +73,19 @@
    private SIPCommander cmder;

    @Autowired
    private SIPCommanderFroPlatform sipCommanderFroPlatform;
    private AudioBroadcastManager audioBroadcastManager;

    @Autowired
    private IDeviceService deviceService;

    @Autowired
    private ISIPCommanderForPlatform sipCommanderFroPlatform;

    @Autowired
    private IRedisCatchStorage redisCatchStorage;

    @Autowired
    private ZLMRTPServerFactory zlmrtpServerFactory;

    @Autowired
    private DeferredResultHolder resultHolder;
@@ -87,12 +105,11 @@
    @Autowired
    private VideoStreamSessionManager streamSession;


    @Autowired
    private IDeviceService deviceService;

    @Autowired
    private UserSetting userSetting;

    @Autowired
    private SipConfig sipConfig;

    @Autowired
    private DynamicTask dynamicTask;
@@ -100,11 +117,25 @@
    @Autowired
    private ZlmHttpHookSubscribe subscribe;

    @Autowired
    private ISIPCommanderForPlatform commanderForPlatform;


    @Qualifier("taskExecutor")
    @Autowired
    private ThreadPoolTaskExecutor taskExecutor;

    @Autowired
    private RedisGbPlayMsgListener redisGbPlayMsgListener;

    @Autowired
    private ZlmHttpHookSubscribe hookSubscribe;


    @Override
    public void play(MediaServerItem mediaServerItem, String deviceId, String channelId,
                                 ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
                                 Runnable timeoutCallback) {
                     ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
                     Runnable timeoutCallback) {
        if (mediaServerItem == null) {
            throw new ControllerException(ErrorCode.ERROR100.getCode(), "未找到可用的zlm");
        }
@@ -165,7 +196,6 @@
                redisCatchStorage.stopPlay(streamInfo);
                storager.stopPlay(streamInfo.getDeviceID(), streamInfo.getChannelId());
                streamInfo = null;

            }
        }
        if (streamInfo == null) {
@@ -213,6 +243,147 @@
        }
    }

    private void talk(MediaServerItem mediaServerItem, Device device, String channelId, String stream,
                      ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
                      Runnable timeoutCallback, AudioBroadcastEvent audioEvent) {

        String playSsrc = mediaServerItem.getSsrcConfig().getPlaySsrc();
        if (playSsrc == null) {
            audioEvent.call("ssrc已经用尽");
            return;
        }
        SendRtpItem sendRtpItem = new SendRtpItem();
        sendRtpItem.setApp("talk");
        sendRtpItem.setStream(stream);
        sendRtpItem.setSsrc(playSsrc);
        sendRtpItem.setDeviceId(device.getDeviceId());
        sendRtpItem.setPlatformId(device.getDeviceId());
        sendRtpItem.setChannelId(channelId);
        sendRtpItem.setRtcp(false);
        sendRtpItem.setMediaServerId(mediaServerItem.getId());
        sendRtpItem.setOnlyAudio(true);
        sendRtpItem.setPlayType(InviteStreamType.TALK);
        sendRtpItem.setPt(8);
        sendRtpItem.setStatus(1);
        sendRtpItem.setTcpActive(false);
        sendRtpItem.setTcp(true);
        sendRtpItem.setUsePs(false);
        sendRtpItem.setReceiveStream(stream + "_talk");


        int port = zlmrtpServerFactory.keepPort(mediaServerItem, playSsrc);
        //端口获取失败的ssrcInfo 没有必要发送点播指令
        if (port <= 0) {
            logger.info("[语音对讲] 端口分配异常，deviceId={},channelId={}", device.getDeviceId(), channelId);
            audioEvent.call("端口分配异常");
            return;
        }
        sendRtpItem.setLocalPort(port);
        sendRtpItem.setPort(port);
        logger.info("[语音对讲]开始 deviceId: {}, channelId: {},收流端口： {}, 收流模式：{}, SSRC: {}, SSRC校验：{}", device.getDeviceId(), channelId, sendRtpItem.getLocalPort(), device.getStreamMode(), sendRtpItem.getSsrc(), false);
        // 超时处理
        String timeOutTaskKey = UUID.randomUUID().toString();
        dynamicTask.startDelay(timeOutTaskKey, () -> {

            logger.info("[语音对讲] 收流超时 deviceId: {}, channelId: {}，端口：{}, SSRC: {}", device.getDeviceId(), channelId, sendRtpItem.getPort(), sendRtpItem.getSsrc());
            timeoutCallback.run();
            // 点播超时回复BYE 同时释放ssrc以及此次点播的资源
            try {
                cmder.streamByeCmd(device, channelId, sendRtpItem.getStream(), null);
            } catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
                logger.error("[语音对讲]超时， 发送BYE失败 {}", e.getMessage());
            } finally {
                timeoutCallback.run();
                mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
                streamSession.remove(device.getDeviceId(), channelId, sendRtpItem.getStream());
            }
        }, userSetting.getPlayTimeout());

        String callId = SipUtils.getNewCallId();

        zlmrtpServerFactory.releasePort(mediaServerItem, playSsrc);
        Map<String, Object> param = new HashMap<>(12);
        param.put("vhost","__defaultVhost__");
        param.put("app", sendRtpItem.getApp());
        param.put("stream", sendRtpItem.getStream());
        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", sendRtpItem.isTcp() ? "0" : "1");
        param.put("recv_stream_id", sendRtpItem.getReceiveStream());
        param.put("close_delay_ms", userSetting.getPlayTimeout() * 1000);

        zlmrtpServerFactory.startSendRtpPassive(mediaServerItem, param, jsonObject -> {
            if (jsonObject == null || jsonObject.getInteger("code") != 0 ) {
                mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
                logger.info("[语音对讲]失败 deviceId: {}, channelId: {}", device.getDeviceId(), channelId);
                audioEvent.call("失败, " + jsonObject.getString("msg"));
                // 查看是否已经建立了通道，存在则发送bye
                stopTalk(device, channelId);
            }
        });


        // 查看设备是否已经在推流
        try {
            cmder.talkStreamCmd(mediaServerItem, sendRtpItem, device, channelId, callId, (MediaServerItem mediaServerItemInuse, JSONObject response) -> {
                logger.info("[语音对讲] 流已生成， 开始推流： " + response.toJSONString());
                dynamicTask.stop(timeOutTaskKey);
                // TODO 暂不做处理
            }, (MediaServerItem mediaServerItemInuse, JSONObject json) -> {
                logger.info("[语音对讲] 设备开始推流： " + json.toJSONString());
                dynamicTask.stop(timeOutTaskKey);

            }, (event) -> {
                dynamicTask.stop(timeOutTaskKey);

                if (event.event instanceof ResponseEvent) {
                    ResponseEvent responseEvent = (ResponseEvent) event.event;
                    if (responseEvent.getResponse() instanceof SIPResponse) {
                        SIPResponse response = (SIPResponse) responseEvent.getResponse();
                        sendRtpItem.setFromTag(response.getFromTag());
                        sendRtpItem.setToTag(response.getToTag());
                        sendRtpItem.setCallId(response.getCallIdHeader().getCallId());
                        redisCatchStorage.updateSendRTPSever(sendRtpItem);

                        streamSession.put(device.getDeviceId(), channelId, "talk",
                                sendRtpItem.getStream(), sendRtpItem.getSsrc(), sendRtpItem.getMediaServerId(),
                                response, VideoStreamSessionManager.SessionType.talk);
                    } else {
                        logger.error("[语音对讲]收到的消息错误，response不是SIPResponse");
                    }
                } else {
                    logger.error("[语音对讲]收到的消息错误，event不是ResponseEvent");
                }

            }, (event) -> {
                dynamicTask.stop(timeOutTaskKey);
                mediaServerService.closeRTPServer(mediaServerItem, sendRtpItem.getStream());
                // 释放ssrc
                mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
                streamSession.remove(device.getDeviceId(), channelId, sendRtpItem.getStream());
                errorEvent.response(event);
            });
        } catch (InvalidArgumentException | SipException | ParseException e) {

            logger.error("[命令发送失败] 对讲消息: {}", e.getMessage());
            dynamicTask.stop(timeOutTaskKey);
            mediaServerService.closeRTPServer(mediaServerItem, sendRtpItem.getStream());
            // 释放ssrc
            mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());

            streamSession.remove(device.getDeviceId(), channelId, sendRtpItem.getStream());
            SipSubscribe.EventResult eventResult = new SipSubscribe.EventResult(new CmdSendFailEvent(null));
            eventResult.msg = "命令发送失败";
            errorEvent.response(eventResult);
        }
//        }

    }



    @Override
    public void play(MediaServerItem mediaServerItem, SSRCInfo ssrcInfo, Device device, String channelId,
@@ -229,7 +400,8 @@
                // 点播超时回复BYE 同时释放ssrc以及此次点播的资源
                try {
                    cmder.streamByeCmd(device, channelId, ssrcInfo.getStream(), null);
                } catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
                } catch (InvalidArgumentException | ParseException | SipException |
                         SsrcTransactionNotFoundException e) {
                    logger.error("[点播超时]， 发送BYE失败 {}", e.getMessage());
                } finally {
                    timeoutCallback.run(1, "收流超时");
@@ -266,7 +438,7 @@
                onPublishHandlerForPlay(mediaServerItemInuse, response, device.getDeviceId(), channelId);
                hookEvent.response(mediaServerItemInuse, response);
                logger.info("[点播成功] deviceId: {}, channelId: {}", device.getDeviceId(), channelId);
                String streamUrl = String.format("http://127.0.0.1:%s/%s/%s.live.flv", mediaServerItemInuse.getHttpPort(), "rtp",  ssrcInfo.getStream());
                String streamUrl = String.format("http://127.0.0.1:%s/%s/%s.live.flv", mediaServerItemInuse.getHttpPort(), "rtp", ssrcInfo.getStream());
                String path = "snap";
                String fileName = device.getDeviceId() + "_" + channelId + ".jpg";
                // 请求截图
@@ -318,7 +490,7 @@
                        // 关闭rtp server
                        mediaServerService.closeRTPServer(mediaServerItem, ssrcInfo.getStream());
                        // 重新开启ssrc server
                        mediaServerService.openRTPServer(mediaServerItem, ssrcInfo.getStream(), ssrcInResponse, device.isSsrcCheck(), false, ssrcInfo.getPort());
                        mediaServerService.openRTPServer(mediaServerItem, ssrcInfo.getStream(), ssrcInResponse, device.isSsrcCheck(), false, ssrcInfo.getPort(), false);

                    }
                }
@@ -435,8 +607,8 @@

    @Override
    public void playBack(String deviceId, String channelId, String startTime,
                                                          String endTime, InviteStreamCallback inviteStreamCallback,
                                                          PlayBackCallback callback) {
                         String endTime, InviteStreamCallback inviteStreamCallback,
                         PlayBackCallback callback) {
        Device device = storager.queryVideoDevice(deviceId);
        if (device == null) {
            return;
@@ -449,9 +621,9 @@

    @Override
    public void playBack(MediaServerItem mediaServerItem, SSRCInfo ssrcInfo,
                                                          String deviceId, String channelId, String startTime,
                                                          String endTime, InviteStreamCallback infoCallBack,
                                                          PlayBackCallback playBackCallback) {
                         String deviceId, String channelId, String startTime,
                         String endTime, InviteStreamCallback infoCallBack,
                         PlayBackCallback playBackCallback) {
        if (mediaServerItem == null || ssrcInfo == null) {
            return;
        }
@@ -559,7 +731,7 @@
                                    // 关闭rtp server
                                    mediaServerService.closeRTPServer(mediaServerItem, ssrcInfo.getStream());
                                    // 重新开启ssrc server
                                    mediaServerService.openRTPServer(mediaServerItem, ssrcInfo.getStream(), ssrcInResponse, device.isSsrcCheck(), true, ssrcInfo.getPort());
                                    mediaServerService.openRTPServer(mediaServerItem, ssrcInfo.getStream(), ssrcInResponse, device.isSsrcCheck(), true, ssrcInfo.getPort(), false);
                                }
                            }
                        }
@@ -573,7 +745,6 @@
            errorEvent.response(eventResult);
        }
    }



    @Override
@@ -760,13 +931,149 @@
                        cmder.streamByeCmd(device, ssrcTransaction.getChannelId(),
                                ssrcTransaction.getStream(), null);
                    } catch (InvalidArgumentException | ParseException | SipException |
                            SsrcTransactionNotFoundException e) {
                             SsrcTransactionNotFoundException e) {
                        logger.error("[zlm离线]为正在使用此zlm的设备， 发送BYE失败 {}", e.getMessage());
                    }
                }
            }
        }
    }

    @Override
    public AudioBroadcastResult audioBroadcast(Device device, String channelId, Boolean broadcastMode) {
        // TODO 必须多端口模式才支持语音喊话鹤语音对讲
        if (device == null || channelId == null) {
            return null;
        }
        logger.info("[语音喊话] device： {}, channel: {}", device.getDeviceId(), channelId);
        DeviceChannel deviceChannel = storager.queryChannel(device.getDeviceId(), channelId);
        if (deviceChannel == null) {
            logger.warn("开启语音广播的时候未找到通道： {}", channelId);
            return null;
        }
        MediaServerItem mediaServerItem = mediaServerService.getMediaServerForMinimumLoad(null);
        if (broadcastMode == null) {
            broadcastMode = true;
        }
        String app = broadcastMode?"broadcast":"talk";
        String stream = device.getDeviceId() + "_" + channelId;
        AudioBroadcastResult audioBroadcastResult = new AudioBroadcastResult();
        audioBroadcastResult.setApp(app);
        audioBroadcastResult.setStream(stream);
        audioBroadcastResult.setStreamInfo(new StreamContent(mediaService.getStreamInfoByAppAndStream(mediaServerItem, app, stream, null, null, null, false)));
        audioBroadcastResult.setCodec("G.711");
        return audioBroadcastResult;
    }

    @Override
    public boolean audioBroadcastCmd(Device device, String channelId, MediaServerItem mediaServerItem, String app, String stream, int timeout, boolean isFromPlatform, AudioBroadcastEvent event) throws InvalidArgumentException, ParseException, SipException {
        if (device == null || channelId == null) {
            return false;
        }
        logger.info("[语音喊话] device： {}, channel: {}", device.getDeviceId(), channelId);
        DeviceChannel deviceChannel = storager.queryChannel(device.getDeviceId(), channelId);
        if (deviceChannel == null) {
            logger.warn("开启语音广播的时候未找到通道： {}", channelId);
            event.call("开启语音广播的时候未找到通道");
            return false;
        }
        // 查询通道使用状态
        if (audioBroadcastManager.exit(device.getDeviceId(), channelId)) {
            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
                Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServerItem, sendRtpItem.getApp(), sendRtpItem.getStream());
                if (streamReady) {
                    logger.warn("语音广播已经开启： {}", channelId);
                    event.call("语音广播已经开启");
                    return false;
                } else {
                    stopAudioBroadcast(device.getDeviceId(), channelId);
                }
            }
        }
        SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(device.getDeviceId(), channelId, null, null);
        if (sendRtpItem != null) {
            MediaServerItem mediaServer = mediaServerService.getOne(sendRtpItem.getMediaServerId());
            Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServer, "rtp", sendRtpItem.getReceiveStream());
            if (streamReady) {
                logger.warn("[语音对讲] 进行中： {}", channelId);
                event.call("语音对讲进行中");
                return false;
            } else {
                stopTalk(device, channelId);
            }
        }

        // 发送通知
        cmder.audioBroadcastCmd(device, channelId, eventResultForOk -> {
            // 发送成功
            AudioBroadcastCatch audioBroadcastCatch = new AudioBroadcastCatch(device.getDeviceId(), channelId, mediaServerItem, app, stream, event, AudioBroadcastCatchStatus.Ready, isFromPlatform);
            audioBroadcastManager.update(audioBroadcastCatch);
        }, eventResultForError -> {
            // 发送失败
            logger.error("语音广播发送失败： {}:{}", channelId, eventResultForError.msg);
            event.call("语音广播发送失败");
            stopAudioBroadcast(device.getDeviceId(), channelId);
        });
        return true;
    }

    @Override
    public boolean audioBroadcastInUse(Device device, String channelId) {
        if (audioBroadcastManager.exit(device.getDeviceId(), channelId)) {
            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 mediaServerServiceOne = mediaServerService.getOne(sendRtpItem.getMediaServerId());
                Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServerServiceOne, sendRtpItem.getApp(), sendRtpItem.getStream());
                if (streamReady) {
                    logger.warn("语音广播通道使用中： {}", channelId);
                    return true;
                }
            }
        }
        return false;
    }


    @Override
    public void stopAudioBroadcast(String deviceId, String channelId) {
        logger.info("[停止对讲] 设备：{}, 通道：{}", deviceId, channelId);
        List<AudioBroadcastCatch> audioBroadcastCatchList = new ArrayList<>();
        if (channelId == null) {
            audioBroadcastCatchList.addAll(audioBroadcastManager.get(deviceId));
        } else {
            audioBroadcastCatchList.add(audioBroadcastManager.get(deviceId, channelId));
        }
        if (audioBroadcastCatchList.size() > 0) {
            for (AudioBroadcastCatch audioBroadcastCatch : audioBroadcastCatchList) {
                Device device = deviceService.getDevice(deviceId);
                if (device == null || audioBroadcastCatch == null) {
                    return;
                }
                SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(deviceId, audioBroadcastCatch.getChannelId(), null, null);
                if (sendRtpItem != null) {
                    redisCatchStorage.deleteSendRTPServer(deviceId, sendRtpItem.getChannelId(), null, null);
                    MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
                    Map<String, Object> param = new HashMap<>();
                    param.put("vhost", "__defaultVhost__");
                    param.put("app", sendRtpItem.getApp());
                    param.put("stream", sendRtpItem.getStream());
                    zlmresTfulUtils.stopSendRtp(mediaInfo, param);
                    try {
                        cmder.streamByeCmd(device, sendRtpItem.getChannelId(), audioBroadcastCatch.getSipTransactionInfo(), null);
                    } catch (InvalidArgumentException | ParseException | SipException |
                             SsrcTransactionNotFoundException e) {
                        logger.error("[消息发送失败] 发送语音喊话BYE失败");
                    }
                }

                audioBroadcastManager.del(deviceId, channelId);
            }
        }
    }


    @Override
    public void zlmServerOnline(String mediaServerId) {
@@ -867,4 +1174,201 @@
        Device device = storager.queryVideoDevice(streamInfo.getDeviceID());
        cmder.playResumeCmd(device, streamInfo);
    }

    @Override
    public void startPushStream(SendRtpItem sendRtpItem, SIPResponse sipResponse, ParentPlatform platform, CallIdHeader callIdHeader) {

        // 开始发流
        // 取消设置的超时任务
//         String channelId = request.getCallIdHeader().getCallId();

        String is_Udp = sendRtpItem.isTcp() ? "0" : "1";
        MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
        logger.info("收到ACK，rtp/{}开始推流, 目标={}:{}，SSRC={}, RTCP={}", sendRtpItem.getStream(),
                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.getStream());
        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", is_Udp);
        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.getStream(),
                    sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isTcp(),
                    sendRtpItem.getLocalPort(), sendRtpItem.getPt(), sendRtpItem.isUsePs(), sendRtpItem.isOnlyAudio());
            redisGbPlayMsgListener.sendMsgForStartSendRtpStream(sendRtpItem.getServerId(), requestPushStreamMsg, json -> {
                startSendRtpStreamHand(sendRtpItem, platform, json, param, callIdHeader);
            });
        } 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) {
                startSendRtpStreamHand(sendRtpItem, platform, startSendRtpStreamResult, param, callIdHeader);
            }
        }
    }

    @Override
    public void startSendRtpStreamHand(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 {
                // 向上级平台
                try {
                    commanderForPlatform.streamByeCmd(parentPlatform, callIdHeader.getCallId());
                } catch (SipException | InvalidArgumentException | ParseException e) {
                    logger.error("[命令发送失败] 国标级联 发送BYE: {}", e.getMessage());
                }
            }
        }
    }

    @Override
    public void talkCmd(Device device, String channelId, MediaServerItem mediaServerItem, String stream, AudioBroadcastEvent event) {
        if (device == null || channelId == null) {
            return;
        }
        // TODO 必须多端口模式才支持语音喊话鹤语音对讲
        logger.info("[语音对讲] device： {}, channel: {}", device.getDeviceId(), channelId);
        DeviceChannel deviceChannel = storager.queryChannel(device.getDeviceId(), channelId);
        if (deviceChannel == null) {
            logger.warn("开启语音对讲的时候未找到通道： {}", channelId);
            event.call("开启语音对讲的时候未找到通道");
            return;
        }
        // 查询通道使用状态
        if (audioBroadcastManager.exit(device.getDeviceId(), channelId)) {
            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 mediaServer = mediaServerService.getOne(sendRtpItem.getMediaServerId());
                Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServer, sendRtpItem.getApp(), sendRtpItem.getStream());
                if (streamReady) {
                    logger.warn("[语音对讲] 正在语音广播，无法开启语音通话： {}", channelId);
                    event.call("正在语音广播");
                    return;
                } else {
                    stopAudioBroadcast(device.getDeviceId(), channelId);
                }
            }
        }

        SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(device.getDeviceId(), channelId, stream, null);
        if (sendRtpItem != null) {
            MediaServerItem mediaServer = mediaServerService.getOne(sendRtpItem.getMediaServerId());
            Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServer, "rtp", sendRtpItem.getReceiveStream());
            if (streamReady) {
                logger.warn("[语音对讲] 进行中： {}", channelId);
                event.call("语音对讲进行中");
                return;
            } else {
                stopTalk(device, channelId);
            }
        }

        talk(mediaServerItem, device, channelId, stream, (MediaServerItem mediaServerItem1, JSONObject response) -> {
            logger.info("[语音对讲] 收到设备发来的流");
        }, eventResult -> {
            logger.warn("[语音对讲] 失败，{}/{}, 错误码 {} {}", device.getDeviceId(), channelId, eventResult.statusCode, eventResult.msg);
            event.call("失败，错误码 " + eventResult.statusCode + ", " + eventResult.msg);
        }, () -> {
            logger.warn("[语音对讲] 失败，{}/{} 超时", device.getDeviceId(), channelId);
            event.call("失败，超时 ");
            stopTalk(device, channelId);
        }, errorMsg -> {
            logger.warn("[语音对讲] 失败，{}/{} {}", device.getDeviceId(), channelId, errorMsg);
            event.call(errorMsg);
            stopTalk(device, channelId);
        });
    }

    private void stopTalk(Device device, String channelId) {
        stopTalk(device, channelId, null);
    }

    @Override
    public void stopTalk(Device device, String channelId, Boolean streamIsReady) {
        logger.info("[语音对讲] 停止， {}/{}", device.getDeviceId(), channelId);
        SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(device.getDeviceId(), channelId, null, null);
        if (sendRtpItem == null) {
            logger.info("[语音对讲] 停止失败， 未找到发送信息，可能已经停止");
            return;
        }
        // 停止向设备推流
        String mediaServerId = sendRtpItem.getMediaServerId();
        if (mediaServerId == null) {
            return;
        }

        MediaServerItem mediaServer = mediaServerService.getOne(mediaServerId);

        if (streamIsReady == null || streamIsReady) {
            Map<String, Object> param = new HashMap<>();
            param.put("vhost", "__defaultVhost__");
            param.put("app", sendRtpItem.getApp());
            param.put("stream", sendRtpItem.getStream());
            param.put("ssrc", sendRtpItem.getSsrc());
            zlmrtpServerFactory.stopSendRtpStream(mediaServer, param);
        }

        mediaServer.getSsrcConfig().releaseSsrc(sendRtpItem.getSsrc());

        SsrcTransaction ssrcTransaction = streamSession.getSsrcTransaction(device.getDeviceId(), channelId, null, sendRtpItem.getStream());
        if (ssrcTransaction != null) {
            try {
                cmder.streamByeCmd(device, channelId, sendRtpItem.getStream(), null);
            } catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException  e) {
                logger.info("[语音对讲] 停止消息发送失败，可能已经停止");
            }
        }
        redisCatchStorage.deleteSendRTPServer(device.getDeviceId(), channelId,null, null);
    }
}