| | |
|---|
| | | 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(), ssrcInfo.getPort());
|
|---|
| | | logger.info("[语音喊话] {} 分配的ZLM为: {} [{}:{}]", stream, mediaServerItem.getId(), mediaServerItem.getIp(), ssrcInfo.getPort());
|
|---|
| | | HookSubscribeForStreamChange hookSubscribeForStreamChange = HookSubscribeFactory.on_stream_changed("rtp", stream, true, "rtsp", mediaServerItem.getId());
|
|---|
| | | subscribe.addSubscribe(hookSubscribeForStreamChange, (MediaServerItem mediaServerItemInUse, JSONObject json) -> {
|
|---|
| | | if (event != null) {
|
|---|
| | |
|---|
| | | content.append("f=v/////a/1/8/1" + "\r\n");
|
|---|
| | |
|
|---|
| | | Request request = headerProvider.createInviteRequest(device, channelId, content.toString(), SipUtils.getNewViaTag(), SipUtils.getNewFromTag(), null, ssrcInfo.getSsrc(), callIdHeader);
|
|---|
| | | sipSender.transmitRequest(device.getTransport(), request, (e -> {
|
|---|
| | | sipSender.transmitRequest(sipLayer.getLocalIp(device.getLocalIp()), request, (e -> {
|
|---|
| | | streamSession.remove(device.getDeviceId(), channelId, ssrcInfo.getStream());
|
|---|
| | | mediaServerService.releaseSsrc(mediaServerItem.getId(), ssrcInfo.getSsrc());
|
|---|
| | | errorEvent.response(e);
|
|---|
