| | |
|---|
| | | return;
|
|---|
| | | }
|
|---|
| | | if (!mediaServerItem.isRtpEnable()) {
|
|---|
| | | // 单端口暂不支持语音对讲
|
|---|
| | | logger.info("[语音对讲] 单端口暂不支持此操作");
|
|---|
| | | // 单端口暂不支持语音喊话
|
|---|
| | | logger.info("[语音喊话] 单端口暂不支持此操作");
|
|---|
| | | return;
|
|---|
| | | }
|
|---|
| | |
|
|---|
| | | logger.info("[语音对讲] {} 分配的ZLM为: {} [{}:{}]", stream, mediaServerItem.getId(), mediaServerItem.getIp(), sendRtpItem.getPort());
|
|---|
| | | logger.info("[语音喊话] {} 分配的ZLM为: {} [{}:{}]", stream, mediaServerItem.getId(), mediaServerItem.getIp(), sendRtpItem.getPort());
|
|---|
| | | HookSubscribeForStreamChange hookSubscribeForStreamChange = HookSubscribeFactory.on_stream_changed("rtp", stream, true, "rtsp", mediaServerItem.getId());
|
|---|
| | | subscribe.addSubscribe(hookSubscribeForStreamChange, (MediaServerItem mediaServerItemInUse, JSONObject json) -> {
|
|---|
| | | if (event != null) {
|
|---|
| | |
|---|
| | | // 这里为例避免一个通道的点播只有一个callID这个参数使用一个固定值
|
|---|
| | | ResponseEvent responseEvent = (ResponseEvent) e.event;
|
|---|
| | | SIPResponse response = (SIPResponse) responseEvent.getResponse();
|
|---|
| | | streamSession.put(device.getDeviceId(), channelId, "talk", stream, sendRtpItem.getSsrc(), mediaServerItem.getId(), response, VideoStreamSessionManager.SessionType.play);
|
|---|
| | | streamSession.put(device.getDeviceId(), channelId, "talk", stream, sendRtpItem.getSsrc(), mediaServerItem.getId(), response, VideoStreamSessionManager.SessionType.talk);
|
|---|
| | | okEvent.response(e);
|
|---|
| | | });
|
|---|
| | | }
|
|---|
