Search results

REST authentication was not supported in earlier versions when the ports 9091 and 8888 was used. For now, just post redirection (return 308 Permanent) is supported. So, to use REST authentication you should use ports 8081 and 8444, as described here.

We sent you a message in reply on your dumps, check the mail

Hello. What WCS version you use? In latest versions, REST API ports are 8081 for HTTP and 8444 for HTTPS. Ports 9091 and 8888 are used to access web interface and sample applications.

Hello. RTSP stream cannot be restored from your dumps. As we can see, your stream is not public. Please, give us access to your stream, our engineers will reproduce the problem on our servers to debug and find solution.

Hello You should not use symbol '@' in username or password because this symbol is reserved and used for splitting credentials and host name. Correct URL rtsp://admin:admin@192.168.11.11/stream1/channel1 Incorrect URL rtsp://admin:ad@min@192.168.11.11/stream1/channel1 As an option you can try to...

Good morning, Marc. Yes, WCS does not support AAC frames of type 0, so, if audio track in file contains only this type, no sound will be heared. Yes, when you play the following file, the new session is created. But mixer output stream will not be interrupted in this case, so you should play...

Добрый день. На вашем сервере запуск WCS занимает около 30 секунд. Если это не так, проверьте, запущен ли haveged ps aux | grep haveged Если не запущен, попробуйте его стартовать haveged или /usr/local/sbin/haveged -w 1024 Если haveged запущен, а проблемы с запуском остаются, остановите /...

Hello We have fixed this issue in build 3502 https://docs.flashphoner.com/display/WCS5EN/Release+notes So it should work properly in latest 3511 too. We found another issue related lost packets. On some rotations recording files can be empty. To avoid this issue you have to add setting...

Добрый день. Проанализировали логи. По какой-то причине от вас приходит битый поток. Портятся сами пакеты данных, которые приходят. 1. С какого браузера тестируете? 2. Какая ОС? 3. Есть ли корпоративные файрволы на пути трафика? Попробуйте этот же тест с другим сервером либо подключиться к...

Hello Marc. We work on RTMP stream playback issue. And yes, /push/sound_on does not mix the sound from file to stream. So, in your case you should use mixer function, by this way: 1. Make a call with /call/startup, "toStream": "stream88" 2. Play an MP4 file with...

Мы взяли дебаговые логи с Вашего сервера, создана внутренняя задача WCS-1449. О результатах сообщим.

Добрый день, Был ли установлен haveged?

Please send output of 'show apps' in CLI on port 2001 and WCS configs (archived dir) and logs to support@flashphoner.com. Or, if possible, send credentials for ssh access to the server.

Hello, In this case, does the stream playback fail, or just video freezes? Please try playing such an RTSP in Chrome in a player on the demo server. If the issue is reproduced, please send network traffic dump of such playback on your server to support@flashphoner.com.

In latest versions REST Hooks do not use RMI and works directly from WCS Core process. So yes you can set disable_manager_rmi=true As a side effect, Monitoring UI will not work. REST Hooks will work.

/call/inject_sound is not suitable for you, because wav file will be heared by callee only. But you can make audio quality better. RTMP stream has PCMA audiotrack, but Player example in browser tries to play PCMU, so quality falls due to transcoding from one to another. To make browser play...

You can stream to WCS 5.1 using WebRTC, RTMP, RTSP protocols. Your hardware must support one of this protocols. Most devices supports RTMP or RTSP. You can perform load test of your server as described here.

Hello The demo server is up and running now. The demo server is used in many scenarios, such as RTSP, RTMP, Transcoding, WebRTC, iOS, Android, etc. It can be stopped by our staff or updated any time. So if you need a stable testing you have to install your own server. If it happens with your...

Good morning, Marc. Let's clarify your case again. If you want to: 1. Make a SIP call 2. Play a wav file to SIP call (and callee should hear this file too) 3. Play a SIP call via WebRTC in browser why don't you use /call/inject_sound REST query, for example and then play stream77 in browser...

We recommend to update WCS version on server B to 5.1.3511. Here is described how to update WCS from 5.0 to 5.1.