优化语音对讲

648540858

2022-11-24 80d96042e7b6f2942585bde482f02a3392477033

 src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/InviteRequestProcessor.java

@@ -11,6 +11,7 @@
import com.genersoft.iot.vmp.gb28181.transmit.SIPProcessorObserver;
import com.genersoft.iot.vmp.gb28181.transmit.SIPSender;
import com.genersoft.iot.vmp.gb28181.transmit.callback.DeferredResultHolder;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.ISIPCommander;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.ISIPCommanderForPlatform;
import com.genersoft.iot.vmp.gb28181.transmit.event.request.ISIPRequestProcessor;
import com.genersoft.iot.vmp.gb28181.transmit.event.request.SIPRequestProcessorParent;
@@ -97,7 +98,7 @@
    private IMediaServerService mediaServerService;

    @Autowired
    private IMediaService mediaService;
    private ISIPCommander commander;

   @Autowired
   private ZLMRESTfulUtils zlmresTfulUtils;
@@ -1003,7 +1004,7 @@
                String stream = device.getDeviceId() + "_" + audioBroadcastCatch.getChannelId();

                CallIdHeader callIdHeader = (CallIdHeader) request.getHeader(CallIdHeader.NAME);
                sendRtpItem.setPlayType(InviteStreamType.PLAY);
                sendRtpItem.setPlayType(InviteStreamType.TALK);
                sendRtpItem.setCallId(callIdHeader.getCallId());
                sendRtpItem.setPlatformId(requesterId);
                sendRtpItem.setStatus(1);
@@ -1017,12 +1018,14 @@

                Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServerItem, app, stream);
                if (streamReady) {
                    SIPResponse sipResponse = sendOk(device, sendRtpItem, sdp, request, mediaServerItem, mediaTransmissionTCP, ssrc);
                    // 添加事务信息
                    streamSession.put(device.getDeviceId(), audioBroadcastCatch.getChannelId(), request.getCallIdHeader().getCallId()
                            , stream,  sendRtpItem.getSsrc(), mediaServerItem.getId(), sipResponse, VideoStreamSessionManager.SessionType.broadcast );
                    sendOk(device, sendRtpItem, sdp, request, mediaServerItem, mediaTransmissionTCP, ssrc);
                }else {
                    logger.warn("[语音通话]， 未发现待推送的流,app={},stream={}", app, stream);
                    try {
                        responseAck(request, Response.GONE);
                    } catch (SipException | InvalidArgumentException | ParseException e) {
                        logger.error("[命令发送失败] 语音通话 回复410失败， {}", e.getMessage());
                    }
                    playService.stopAudioBroadcast(device.getDeviceId(), audioBroadcastCatch.getChannelId());
                }
            } catch (SdpException e) {