| | |
| | | import com.genersoft.iot.vmp.conf.exception.SsrcTransactionNotFoundException; |
| | | import com.genersoft.iot.vmp.gb28181.bean.*; |
| | | import com.genersoft.iot.vmp.gb28181.event.EventPublisher; |
| | | import com.genersoft.iot.vmp.gb28181.session.AudioBroadcastManager;
|
| | | import com.genersoft.iot.vmp.gb28181.event.subscribe.catalog.CatalogEvent; |
| | | import com.genersoft.iot.vmp.gb28181.session.SSRCFactory; |
| | | 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.media.zlm.dto.HookType; |
| | | import com.genersoft.iot.vmp.media.zlm.dto.MediaServerItem; |
| | | import com.genersoft.iot.vmp.media.zlm.dto.StreamAuthorityInfo; |
| | |
| | | private SIPCommander cmder; |
| | | |
| | | @Autowired |
| | | private SIPCommanderFroPlatform commanderFroPlatform; |
| | | private ISIPCommanderForPlatform commanderFroPlatform;
|
| | |
|
| | | @Autowired
|
| | | private AudioBroadcastManager audioBroadcastManager;
|
| | |
|
| | | @Autowired
|
| | | private ZLMServerFactory zlmServerFactory;
|
| | | |
| | | @Autowired |
| | | private IPlayService playService; |
| | |
| | | result.setModify_stamp(2); |
| | | } |
| | | } |
| | | // 如果是talk对讲,则默认获取声音
|
| | | if (ssrcTransactionForAll.get(0).getType() == InviteSessionType.TALK) {
|
| | | result.setEnable_audio(true);
|
| | | } |
| | | }
|
| | | } else if (param.getApp().equals("broadcast")) {
|
| | | result.setEnable_audio(true);
|
| | | } else if (param.getApp().equals("talk")) {
|
| | | result.setEnable_audio(true);
|
| | | } |
| | | if (param.getApp().equalsIgnoreCase("rtp")) { |
| | | String receiveKey = VideoManagerConstants.WVP_OTHER_RECEIVE_RTP_INFO + userSetting.getServerId() + "_" + param.getStream(); |
| | |
| | | } else { |
| | | logger.info("[ZLM HOOK] 流注销, {}->{}->{}/{}", param.getMediaServerId(), param.getSchema(), param.getApp(), param.getStream()); |
| | | } |
| | | |
| | | |
| | | JSONObject json = (JSONObject) JSON.toJSON(param); |
| | | taskExecutor.execute(() -> { |
| | |
| | | redisCatchStorage.updateStreamAuthorityInfo(param.getApp(), param.getStream(), streamAuthorityInfo); |
| | | } |
| | | } |
| | | |
| | | if ("rtsp".equals(param.getSchema())) { |
| | | // 更新流媒体负载信息 |
| | | logger.info("流变化:注册->{}, app->{}, stream->{}", param.isRegist(), param.getApp(), param.getStream());
|
| | | if (param.isRegist()) { |
| | | mediaServerService.addCount(param.getMediaServerId()); |
| | | } else { |
| | | mediaServerService.removeCount(param.getMediaServerId()); |
| | | } |
| | | // 设置拉流代理上线/离线 |
| | |
|
| | | int updateStatusResult = streamProxyService.updateStatus(param.isRegist(), param.getApp(), param.getStream()); |
| | | if (updateStatusResult > 0) { |
| | | |
| | |
| | | inviteStreamService.removeInviteInfo(inviteInfo); |
| | | storager.stopPlay(inviteInfo.getDeviceId(), inviteInfo.getChannelId()); |
| | | } |
| | | } else if ("broadcast".equals(param.getApp())) {
|
| | | // 语音对讲推流 stream需要满足格式deviceId_channelId
|
| | | if (param.getStream().indexOf("_") > 0) {
|
| | | String[] streamArray = param.getStream().split("_");
|
| | | if (streamArray.length == 2) {
|
| | | String deviceId = streamArray[0];
|
| | | String channelId = streamArray[1];
|
| | | Device device = deviceService.getDevice(deviceId);
|
| | | if (device != null) {
|
| | | if (param.isRegist()) {
|
| | | if (audioBroadcastManager.exit(deviceId, channelId)) {
|
| | | playService.stopAudioBroadcast(deviceId, channelId);
|
| | | }
|
| | | // 开启语音对讲通道
|
| | | try {
|
| | | playService.audioBroadcastCmd(device, channelId, mediaInfo, param.getApp(), param.getStream(), 60, false, (msg) -> {
|
| | | logger.info("[语音对讲] 通道建立成功, device: {}, channel: {}", deviceId, channelId);
|
| | | });
|
| | | } catch (InvalidArgumentException | ParseException | SipException e) {
|
| | | logger.error("[命令发送失败] 语音对讲: {}", e.getMessage());
|
| | | }
|
| | | } else {
|
| | | // 流注销
|
| | | playService.stopAudioBroadcast(deviceId, channelId);
|
| | | }
|
| | | } else {
|
| | | logger.info("[语音对讲] 未找到设备:{}", deviceId);
|
| | | }
|
| | | }
|
| | | }
|
| | | } else if ("talk".equals(param.getApp())) {
|
| | | // 语音对讲推流 stream需要满足格式deviceId_channelId
|
| | | if (param.getStream().indexOf("_") > 0) {
|
| | | String[] streamArray = param.getStream().split("_");
|
| | | if (streamArray.length == 2) {
|
| | | String deviceId = streamArray[0];
|
| | | String channelId = streamArray[1];
|
| | | Device device = deviceService.getDevice(deviceId);
|
| | | if (device != null) {
|
| | | if (param.isRegist()) {
|
| | | if (audioBroadcastManager.exit(deviceId, channelId)) {
|
| | | playService.stopAudioBroadcast(deviceId, channelId);
|
| | | }
|
| | | // 开启语音对讲通道
|
| | | playService.talkCmd(device, channelId, mediaInfo, param.getStream(), (msg) -> {
|
| | | logger.info("[语音对讲] 通道建立成功, device: {}, channel: {}", deviceId, channelId);
|
| | | });
|
| | | } else {
|
| | | // 流注销
|
| | | playService.stopTalk(device, channelId, param.isRegist());
|
| | | }
|
| | | } else {
|
| | | logger.info("[语音对讲] 未找到设备:{}", deviceId);
|
| | | }
|
| | | }
|
| | | }
|
| | |
|
| | | } else { |
| | | if (!"rtp".equals(param.getApp())) { |
| | | String type = OriginType.values()[param.getOriginType()].getType(); |
| | |
| | | if (platform != null) { |
| | | commanderFroPlatform.streamByeCmd(platform, sendRtpItem); |
| | | redisCatchStorage.deleteSendRTPServer(platformId, sendRtpItem.getChannelId(), |
| | | sendRtpItem.getCallId(), sendRtpItem.getStreamId()); |
| | | sendRtpItem.getCallId(), sendRtpItem.getStream());
|
| | | } else { |
| | | cmder.streamByeCmd(device, sendRtpItem.getChannelId(), param.getStream(), sendRtpItem.getCallId()); |
| | | if (sendRtpItem.getPlayType().equals(InviteStreamType.BROADCAST)
|
| | | || sendRtpItem.getPlayType().equals(InviteStreamType.TALK)) {
|
| | | AudioBroadcastCatch audioBroadcastCatch = audioBroadcastManager.get(sendRtpItem.getDeviceId(), sendRtpItem.getChannelId());
|
| | | if (audioBroadcastCatch != null) {
|
| | | // 来自上级平台的停止对讲
|
| | | logger.info("[停止对讲] 来自上级,平台:{}, 通道:{}", sendRtpItem.getDeviceId(), sendRtpItem.getChannelId());
|
| | | audioBroadcastManager.del(sendRtpItem.getDeviceId(), sendRtpItem.getChannelId());
|
| | | }
|
| | | }
|
| | | } |
| | | } catch (SipException | InvalidArgumentException | ParseException | |
| | | SsrcTransactionNotFoundException e) { |
| | | logger.error("[命令发送失败] 国标级联 发送BYE: {}", e.getMessage()); |
| | | logger.error("[命令发送失败] 发送BYE: {}", e.getMessage());
|
| | | } |
| | | } |
| | | } |
| | |
| | | } |
| | | } |
| | | }); |
| | | |
| | | return HookResult.SUCCESS(); |
| | | } |
| | | |
| | |
| | | logger.error("[命令发送失败] 国标级联 发送BYE: {}", e.getMessage()); |
| | | } |
| | | redisCatchStorage.deleteSendRTPServer(parentPlatform.getServerGBId(), sendRtpItem.getChannelId(), |
| | | sendRtpItem.getCallId(), sendRtpItem.getStreamId()); |
| | | sendRtpItem.getCallId(), sendRtpItem.getStream());
|
| | | if (InviteStreamType.PUSH == sendRtpItem.getPlayType()) { |
| | | MessageForPushChannel messageForPushChannel = MessageForPushChannel.getInstance(0, |
| | | sendRtpItem.getApp(), sendRtpItem.getStreamId(), sendRtpItem.getChannelId(), |
| | | sendRtpItem.getApp(), sendRtpItem.getStream(), sendRtpItem.getChannelId(),
|
| | | sendRtpItem.getPlatformId(), parentPlatform.getName(), userSetting.getServerId(), sendRtpItem.getMediaServerId()); |
| | | messageForPushChannel.setPlatFormIndex(parentPlatform.getId()); |
| | | redisCatchStorage.sendPlatformStopPlayMsg(messageForPushChannel); |
| | |
| | | storager.stopPlay(inviteInfo.getDeviceId(), inviteInfo.getChannelId()); |
| | | return ret; |
| | | } |
| | | SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(null, null, param.getStream(), null);
|
| | | if (sendRtpItem != null && "talk".equals(sendRtpItem.getApp())) {
|
| | | ret.put("close", false);
|
| | | return ret;
|
| | | }
|
| | | } else if ("talk".equals(param.getApp()) || "broadcast".equals(param.getApp())) {
|
| | | ret.put("close", false);
|
| | | } else { |
| | | // 非国标流 推流/拉流代理 |
| | | // 拉流代理 |
| | |
| | | |
| | | if (!exist) { |
| | | SSRCInfo ssrcInfo = mediaServerService.openRTPServer(mediaInfo, param.getStream(), null, |
| | | device.isSsrcCheck(), true, 0, false, device.getStreamModeForParam()); |
| | | device.isSsrcCheck(), true, 0, false, false, device.getStreamModeForParam());
|
| | | playService.playBack(mediaInfo, ssrcInfo, deviceId, channelId, startTime, endTime, (code, message, data) -> { |
| | | msg.setData(new HookResult(code, message)); |
| | | resultHolder.invokeResult(msg); |
| | |
| | | logger.error("[命令发送失败] 国标级联 发送BYE: {}", e.getMessage()); |
| | | } |
| | | redisCatchStorage.deleteSendRTPServer(parentPlatform.getServerGBId(), sendRtpItem.getChannelId(), |
| | | sendRtpItem.getCallId(), sendRtpItem.getStreamId()); |
| | | sendRtpItem.getCallId(), sendRtpItem.getStream());
|
| | | } |
| | | } |
| | | }); |
| | |
| | | */ |
| | | @ResponseBody |
| | | @PostMapping(value = "/on_rtp_server_timeout", produces = "application/json;charset=UTF-8") |
| | | public HookResult onRtpServerTimeout(HttpServletRequest request, @RequestBody OnRtpServerTimeoutHookParam param) { |
| | | public HookResult onRtpServerTimeout(HttpServletRequest request, @RequestBody OnRtpServerTimeoutHookParam
|
| | | param) {
|
| | | logger.info("[ZLM HOOK] rtpServer收流超时:{}->{}({})", param.getMediaServerId(), param.getStream_id(), param.getSsrc()); |
| | | |
| | | taskExecutor.execute(() -> { |