| | |
| | | import com.genersoft.iot.vmp.conf.UserSetting; |
| | | import com.genersoft.iot.vmp.conf.VersionInfo; |
| | | import com.genersoft.iot.vmp.conf.exception.ControllerException; |
| | | import com.genersoft.iot.vmp.gb28181.bean.SendRtpItem; |
| | | import com.genersoft.iot.vmp.media.zlm.ZlmHttpHookSubscribe; |
| | | import com.genersoft.iot.vmp.media.zlm.dto.MediaServerItem; |
| | | import com.genersoft.iot.vmp.service.*; |
| | |
| | | private IRedisCatchStorage redisCatchStorage; |
| | | |
| | | |
| | | @GetMapping(value = "/openRtpServer") |
| | | @GetMapping(value = "/receive/open") |
| | | @ResponseBody |
| | | @Operation(summary = "开启收流和获取发流信息") |
| | | @Parameter(name = "isSend", description = "是否发送,false时同时只开启收流", required = true) |
| | | @Parameter(name = "callId", description = "整个过程的唯一标识", required = true) |
| | | @Parameter(name = "ssrc", description = "来源流的SSRC", required = false) |
| | | @Parameter(name = "hasAudio", description = "是否", required = false) |
| | | @Parameter(name = "isSend", description = "是否发送,false时只开启收流, true同时返回推流信息", required = true) |
| | | @Parameter(name = "callId", description = "整个过程的唯一标识,为了与后续接口关联", required = true) |
| | | @Parameter(name = "ssrc", description = "来源流的SSRC,不传则不校验来源ssrc", required = false) |
| | | @Parameter(name = "stream", description = "形成的流的ID", required = true) |
| | | @Parameter(name = "tcpMode", description = "收流模式, 0为UDP, 1为TCP被动", required = true) |
| | | public void openRtpServer(Boolean isSend, String ssrc, String callId, Boolean hasAudio, String stream, Integer tcpMode) { |
| | | @Parameter(name = "callBack", description = "回调地址,如果收流超时会通道回调通知,回调为get请求,参数为callId", required = true) |
| | | public SendRtpItem openRtpServer(Boolean isSend, String ssrc, String callId, String stream, Integer tcpMode, String callBack) { |
| | | MediaServerItem mediaServerItem = mediaServerService.getMediaServerForMinimumLoad(null); |
| | | if (mediaServerItem == null) { |
| | | throw new ControllerException(ErrorCode.ERROR100.getCode(),"没有可用的MediaServer"); |
| | | } |
| | | return null; |
| | | } |
| | | |
| | | @GetMapping(value = "/sendRTP") |
| | | @GetMapping(value = "/receive/close") |
| | | @ResponseBody |
| | | @Operation(summary = "关闭收流") |
| | | @Parameter(name = "stream", description = "流的ID", required = true) |
| | | public void closeRtpServer(String stream) { |
| | | |
| | | } |
| | | |
| | | @GetMapping(value = "/send/start") |
| | | @ResponseBody |
| | | @Operation(summary = "发送流") |
| | | |
| | | @Parameter(name = "ssrc", description = "发送流的SSRC", required = true) |
| | | @Parameter(name = "ip", description = "目标IP", required = true) |
| | | @Parameter(name = "port", description = "目标端口", required = true) |
| | | @Parameter(name = "app", description = "待发送应用名", required = true) |
| | | @Parameter(name = "stream", description = "待发送流Id", required = true) |
| | | @Parameter(name = "callId", description = "整个过程的唯一标识", required = true) |
| | | @Parameter(name = "callId", description = "整个过程的唯一标识,不传则使用随机端口发流", required = true) |
| | | @Parameter(name = "onlyAudio", description = "是否只有音频", required = true) |
| | | public void sendRTP(String ssrc, String ip, Integer port, String app, String stream, String callId, Boolean onlyAudio) { |
| | | @Parameter(name = "streamType", description = "流类型,1为es流,2为ps流, 默认es流", required = false) |
| | | public void sendRTP(String ssrc, String ip, Integer port, String app, String stream, String callId, Boolean onlyAudio, Integer streamType) { |
| | | |
| | | } |
| | | |
| | | |
| | | |
| | | @GetMapping(value = "/send/stop") |
| | | @ResponseBody |
| | | @Operation(summary = "关闭发送流") |
| | | @Parameter(name = "callId", description = "整个过程的唯一标识,不传则使用随机端口发流", required = true) |
| | | public void closeSendRTP(String callId) { |
| | | |
| | | } |
| | | |