添加第二种语音对讲实现

648540858

2022-09-29 06bbe3fe01e5af9486c309693a975077df813f7c

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

@@ -12,8 +12,10 @@
import com.genersoft.iot.vmp.gb28181.bean.*;
import com.genersoft.iot.vmp.conf.exception.ControllerException;
import com.genersoft.iot.vmp.conf.exception.SsrcTransactionNotFoundException;
import com.genersoft.iot.vmp.gb28181.bean.*;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.ISIPCommanderForPlatform;
import com.genersoft.iot.vmp.gb28181.utils.SipUtils;
import com.genersoft.iot.vmp.service.IDeviceService;
import com.genersoft.iot.vmp.vmanager.bean.AudioBroadcastResult;
import com.genersoft.iot.vmp.vmanager.bean.ErrorCode;
import org.slf4j.Logger;
import org.slf4j.LoggerFactory;
@@ -36,7 +38,6 @@
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.impl.SIPCommander;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.impl.SIPCommanderFroPlatform;
import com.genersoft.iot.vmp.media.zlm.ZLMRTPServerFactory;
import com.genersoft.iot.vmp.media.zlm.dto.HookSubscribeFactory;
import com.genersoft.iot.vmp.media.zlm.dto.HookSubscribeForStreamChange;
@@ -54,32 +55,9 @@
import com.genersoft.iot.vmp.service.bean.SSRCInfo;
import com.genersoft.iot.vmp.storager.IRedisCatchStorage;
import com.genersoft.iot.vmp.storager.IVideoManagerStorage;
import com.genersoft.iot.vmp.utils.redis.RedisUtil;
import com.genersoft.iot.vmp.vmanager.bean.AudioBroadcastResult;
import com.genersoft.iot.vmp.utils.DateUtil;
import com.genersoft.iot.vmp.vmanager.bean.WVPResult;
import com.genersoft.iot.vmp.vmanager.gb28181.play.bean.AudioBroadcastEvent;
import com.genersoft.iot.vmp.vmanager.gb28181.play.bean.PlayResult;

import gov.nist.javax.sip.stack.SIPDialog;
import org.slf4j.Logger;
import org.slf4j.LoggerFactory;
import org.springframework.beans.factory.annotation.Autowired;
import org.springframework.http.HttpStatus;
import org.springframework.http.ResponseEntity;
import org.springframework.stereotype.Service;
import org.springframework.util.ResourceUtils;
import org.springframework.web.context.request.async.DeferredResult;

import javax.sip.ResponseEvent;
import javax.sip.SipException;
import java.io.FileNotFoundException;
import java.math.BigDecimal;
import java.text.ParseException;
import java.math.RoundingMode;
import java.util.*;
import java.util.stream.Collectors;
import java.util.stream.Stream;

@SuppressWarnings(value = {"rawtypes", "unchecked"})
@Service
@@ -97,7 +75,10 @@
    private AudioBroadcastManager audioBroadcastManager;

    @Autowired
    private SIPCommanderFroPlatform sipCommanderFroPlatform;
    private IDeviceService deviceService;

    @Autowired
    private ISIPCommanderForPlatform sipCommanderFroPlatform;

    @Autowired
    private IRedisCatchStorage redisCatchStorage;
@@ -123,10 +104,6 @@
    @Autowired
    private VideoStreamSessionManager streamSession;


    @Autowired
    private IDeviceService deviceService;

    @Autowired
    private UserSetting userSetting;

@@ -143,7 +120,6 @@
    @Qualifier("taskExecutor")
    @Autowired
    private ThreadPoolTaskExecutor taskExecutor;



    @Override
@@ -277,6 +253,186 @@
        return playResult;
    }

    @Override
    public void talk(MediaServerItem mediaServerItem, Device device, String channelId,
                     ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
                     Runnable timeoutCallback) {
        String streamId = null;
        if (mediaServerItem.isRtpEnable()) {
            streamId = String.format("%s_%s", device.getDeviceId(), channelId);
        }
        SSRCInfo ssrcInfo = mediaServerService.openRTPServer(mediaServerItem, streamId, device.isSsrcCheck(), false);
        logger.info("[对讲开始] deviceId: {}, channelId: {},收流端口： {}, 收流模式：{}, SSRC: {}, SSRC校验：{}", device.getDeviceId(), channelId, ssrcInfo.getPort(), device.getStreamMode(), ssrcInfo.getSsrc(), device.isSsrcCheck());
        // 超时处理
        String timeOutTaskKey = UUID.randomUUID().toString();
        SSRCInfo finalSsrcInfo = ssrcInfo;
        System.out.println("设置超时任务： " + timeOutTaskKey);
        dynamicTask.startDelay(timeOutTaskKey, () -> {

            logger.info("[对讲超时] 收流超时 deviceId: {}, channelId: {}，端口：{}, SSRC: {}", device.getDeviceId(), channelId, finalSsrcInfo.getPort(), finalSsrcInfo.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, finalSsrcInfo.getStream(), null);
            } catch (InvalidArgumentException | ParseException | SipException e) {
                logger.error("[对讲超时]， 发送BYE失败 {}", e.getMessage());
            } catch (SsrcTransactionNotFoundException e) {
                timeoutCallback.run();
                mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
                mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
                streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
            }
        }, userSetting.getPlayTimeout());
        final String ssrc = ssrcInfo.getSsrc();
        final String stream = ssrcInfo.getStream();
        //端口获取失败的ssrcInfo 没有必要发送点播指令
        if (ssrcInfo.getPort() <= 0) {
            logger.info("[对讲] 端口分配异常，deviceId={},channelId={},ssrcInfo={}", device.getDeviceId(), channelId, ssrcInfo);
            return;
        }
        try {
            String callId = SipUtils.getNewCallId();
            cmder.talkStreamCmd(mediaServerItem, ssrcInfo, device, channelId, callId, (MediaServerItem mediaServerItemInuse, JSONObject response) -> {
                logger.info("[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("[对讲] 开始推流： " + json.toJSONString());
                dynamicTask.stop(timeOutTaskKey);
                // 获取远程IP端口 作为回复语音流的地址
                String ip = json.getString("ip");
                Integer port = json.getInteger("port");
                logger.info("[远端设备开始推流]{}/{}, 来自ip：{}, 端口：{}", device.getDeviceId(), channelId, ip, port);
                // 查看平台推流是否就绪
                Boolean ready = zlmrtpServerFactory.isStreamReady(mediaServerItemInuse, "talk", stream);
                if (!ready) {
                    try {
                        cmder.streamByeCmd(device, channelId, finalSsrcInfo.getStream(), null);
                    } catch (InvalidArgumentException | ParseException | SipException e) {
                        logger.error("[对讲超时]， 发送BYE失败 {}", e.getMessage());
                    } catch (SsrcTransactionNotFoundException e) {
                        timeoutCallback.run();
                        mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
                        mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
                        streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
                    }
                }else {
                    SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, ip, port, ssrcInfo.getSsrc(), device.getDeviceId(),
                            device.getDeviceId(), channelId,
                            false);

                    sendRtpItem.setTcpActive(false);
                    if (sendRtpItem == null || sendRtpItem.getLocalPort() == 0) {
                        logger.warn("服务器端口资源不足");
                        try {
                            cmder.streamByeCmd(device, channelId, finalSsrcInfo.getStream(), null);
                        } catch (InvalidArgumentException | ParseException | SipException e) {
                            logger.error("[对讲超时]， 发送BYE失败 {}", e.getMessage());
                        } catch (SsrcTransactionNotFoundException e) {
                            timeoutCallback.run();
                            mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
                            mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
                            streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
                        }
                        return;
                    }
                    sendRtpItem.setCallId(callId);
                    sendRtpItem.setPlayType(InviteStreamType.TALK);
                    sendRtpItem.setStatus(1);
                    sendRtpItem.setIp(ip);
                    sendRtpItem.setPort(port);
                    sendRtpItem.setTcpActive(false);
                    sendRtpItem.setStreamId(ssrcInfo.getStream());
                    sendRtpItem.setApp("talk");
                    sendRtpItem.setSsrc(ssrc);
                    redisCatchStorage.updateSendRTPSever(sendRtpItem);

                    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");
                    JSONObject jsonObject = zlmrtpServerFactory.startSendRtpStream(mediaServerItemInuse, param);
                    System.out.println(11111);
                    System.out.println(jsonObject);
                }

            }, (event) -> {
//                ResponseEvent responseEvent = (ResponseEvent) event.event;
//                String contentString = new String(responseEvent.getResponse().getRawContent());
//                // 获取ssrc
//                int ssrcIndex = contentString.indexOf("y=");
//                // 检查是否有y字段
//                if (ssrcIndex >= 0) {
//                    //ssrc规定长度为10字节，不取余下长度以避免后续还有“f=”字段 TODO 后续对不规范的非10位ssrc兼容
//                    String ssrcInResponse = contentString.substring(ssrcIndex + 2, ssrcIndex + 12);
//                    // 查询到ssrc不一致且开启了ssrc校验则需要针对处理
//                    if (ssrc.equals(ssrcInResponse)) {
//                        return;
//                    }
//                    logger.info("[对讲消息] 收到invite 200, 发现下级自定义了ssrc: {}", ssrcInResponse);
//                    if (!mediaServerItem.isRtpEnable() || device.isSsrcCheck()) {
//                        logger.info("[对讲消息] SSRC修正 {}->{}", ssrc, ssrcInResponse);
//
//                        if (!mediaServerItem.getSsrcConfig().checkSsrc(ssrcInResponse)) {
//                            // ssrc 不可用
//                            // 释放ssrc
//                            mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
//                            streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
//                            event.msg = "下级自定义了ssrc,但是此ssrc不可用";
//                            event.statusCode = 400;
//                            errorEvent.response(event);
//                            return;
//                        }
//
//                        // 单端口模式streamId也有变化，需要重新设置监听
//                        if (!mediaServerItem.isRtpEnable()) {
//                            // 添加订阅
//                            HookSubscribeForStreamChange hookSubscribe = HookSubscribeFactory.on_stream_changed("rtp", stream, true, "rtsp", mediaServerItem.getId());
//                            subscribe.removeSubscribe(hookSubscribe);
//                            hookSubscribe.getContent().put("stream", String.format("%08x", Integer.parseInt(ssrcInResponse)).toUpperCase());
//                            subscribe.addSubscribe(hookSubscribe, (MediaServerItem mediaServerItemInUse, JSONObject response) -> {
//                                logger.info("[ZLM HOOK] ssrc修正后收到订阅消息： " + response.toJSONString());
//                                dynamicTask.stop(timeOutTaskKey);
//                                // hook响应
//                                onPublishHandlerForPlay(mediaServerItemInUse, response, device.getDeviceId(), channelId, uuid);
//                                hookEvent.response(mediaServerItemInUse, response);
//                            });
//                        }
//                        // 关闭rtp server
//                        mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
//                        // 重新开启ssrc server
//                        mediaServerService.openRTPServer(mediaServerItem, finalSsrcInfo.getStream(), ssrcInResponse, device.isSsrcCheck(), false, finalSsrcInfo.getPort());
//
//                    }
//                }
            }, (event) -> {
                dynamicTask.stop(timeOutTaskKey);
                mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
                // 释放ssrc
                mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());

                streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.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, finalSsrcInfo.getStream());
            // 释放ssrc
            mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());

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


    @Override
@@ -806,10 +962,36 @@
    }

    @Override
    public void audioBroadcast(Device device, String channelId, int timeout, AudioBroadcastEvent event) {
    public AudioBroadcastResult audioBroadcast(Device device, String channelId) {
        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();
//        String app = "broadcast";
        // TODO 从sip user agent中判断是什么品牌设备，大华默认使用talk模式，其他使用broadcast模式
        String app = "talk";
        String stream = device.getDeviceId() + "_" + channelId;
        StreamInfo broadcast = mediaService.getStreamInfoByAppAndStream(mediaServerItem, "broadcast", stream, null, null, null, false);
        AudioBroadcastResult audioBroadcastResult = new AudioBroadcastResult();
        audioBroadcastResult.setApp(app);
        audioBroadcastResult.setStream(stream);
        audioBroadcastResult.setStreamInfo(mediaService.getStreamInfoByAppAndStream(mediaServerItem, app, stream, null, null, null,false));
        audioBroadcastResult.setCodec("G.711");
        return audioBroadcastResult;
    }

    @Override
    public void audioBroadcastCmd(Device device, String channelId, int timeout, AudioBroadcastEvent event) throws InvalidArgumentException, ParseException, SipException {
        if (device == null || channelId == null) {
            return;
        }
        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);
@@ -847,12 +1029,20 @@
        });
    }



    @Override
    public void stopAudioBroadcast(String deviceId, String channelId){
        AudioBroadcastCatch audioBroadcastCatch = audioBroadcastManager.get(deviceId, channelId);
        if (audioBroadcastCatch != null) {

            try {
            Device device = deviceService.queryDevice(deviceId);
            if (device == null) {
                return;
            }
//            if (audioBroadcastCatch.getStatus() == AudioBroadcastCatchStatus.Ok) {
//                cmder.streamByeCmd(device, audioBroadcastCatch.getChannelId(), null, audioBroadcastCatch.getSipTransactionInfo().getCallId());
//            }
                SendRtpItem sendRtpItem =  redisCatchStorage.querySendRTPServer(deviceId, audioBroadcastCatch.getChannelId(), null, null);
                if (sendRtpItem != null) {
                    redisCatchStorage.deleteSendRTPServer(deviceId, sendRtpItem.getChannelId(), null, null);
@@ -862,24 +1052,10 @@
                    param.put("app", sendRtpItem.getApp());
                    param.put("stream", sendRtpItem.getStreamId());
                    zlmresTfulUtils.stopSendRtp(mediaInfo, param);
                    // 立刻结束设备的推流，等待自行结束太慢
                    zlmresTfulUtils.closeStreams(mediaInfo, sendRtpItem.getApp(), sendRtpItem.getStreamId());
                }
                if (audioBroadcastCatch.getStatus() == AudioBroadcastCatchStatus.Ok) {
                    cmder.streamByeCmd(audioBroadcastCatch.getDialog(), audioBroadcastCatch.getChannelId(), audioBroadcastCatch.getRequest(), null);
                }

                audioBroadcastManager.del(deviceId, channelId);

            } catch (SipException e) {
                throw new RuntimeException(e);
            } catch (ParseException e) {
                throw new RuntimeException(e);
            } catch (InvalidArgumentException e) {
                throw new RuntimeException(e);
            }
        }


    }

    @Override