| | |
| | | import com.genersoft.iot.vmp.gb28181.transmit.event.request.SIPRequestProcessorParent; |
| | | 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.MediaServerItem; |
| | | import com.genersoft.iot.vmp.service.IMediaServerService; |
| | | import com.genersoft.iot.vmp.service.IPlayService; |
| | |
| | | import javax.sip.header.FromHeader; |
| | | import javax.sip.header.HeaderAddress; |
| | | import javax.sip.header.ToHeader; |
| | | import java.text.ParseException; |
| | | import java.util.HashMap; |
| | | import java.util.Map; |
| | | |
| | |
| | | logger.warn("[收到ACK]:未找到通道({})的推流信息", channelId); |
| | | return; |
| | | } |
| | | String is_Udp = sendRtpItem.isTcp() ? "0" : "1"; |
| | | String isUdp = sendRtpItem.isTcp() ? "0" : "1"; |
| | | MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId()); |
| | | logger.info("收到ACK,rtp/{}开始向上级推流, 目标={}:{},SSRC={}, RTCP={}", sendRtpItem.getStreamId(), |
| | | 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.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"); |
| | | param.put("is_udp", isUdp); |
| | | 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.getStreamId(), |
| | | sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isTcp(), |
| | | sendRtpItem.getLocalPort(), sendRtpItem.getPt(), sendRtpItem.isUsePs(), sendRtpItem.isOnlyAudio()); |
| | | redisGbPlayMsgListener.sendMsgForStartSendRtpStream(sendRtpItem.getServerId(), requestPushStreamMsg, json -> { |
| | | startSendRtpStreamHand(evt, sendRtpItem, parentPlatform, json, callIdHeader); |
| | | playService.startSendRtpStreamHand(sendRtpItem, parentPlatform, json, param, callIdHeader); |
| | | }); |
| | | }else { |
| | | JSONObject startSendRtpStreamResult = zlmrtpServerFactory.startSendRtp(mediaInfo, sendRtpItem); |
| | | if (startSendRtpStreamResult != null) { |
| | | startSendRtpStreamHand(evt, sendRtpItem, parentPlatform, startSendRtpStreamResult, callIdHeader); |
| | | } |
| | | } |
| | | } |
| | | } |
| | | private void startSendRtpStreamHand(RequestEvent evt, 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 { |
| | | // 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) { |
| | | playService.startSendRtpStreamHand(sendRtpItem, parentPlatform, startSendRtpStreamResult, param, callIdHeader); |
| | | } |
| | | } |
| | | } |
| | | |
| | | } |
| | | |
| | | } |