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Мы обновили на Вашем сервере версию WCS до последней на текущий момент 5.1.3511 и установили параметры rtp_receive_buffer_size=131072 rtp_send_buffer_size =131072 т.к. проблемы наблюдаются не с записью потока, а с его публикацией, в какой-то момент переполняется буфер входящих UDP-пакетов...

В версиях 5.1.3511 и 5.0.3504 добавили REST метод, который позволяет включить дебаг логи для конкретной сессии. Документация здесь. Если есть проблемы с зеленым экраном или отображением, попробуйте поставить эту сессию на дебаг и снять логи.

Hello Try to disable manager RMI disable_manager_rmi=true It should prevent errors in flashphoner_manager.log Regarding error. If you can reproduce this, try to update to the latest version 3511 https://docs.flashphoner.com/display/WCS5EN/Release+notes and enable debug log on user's session...

Пришлите полный файл лога при котором воспроизводится эта проблема. support@flashphoner.com Эта ошибка говорит о том что UDP порт занят. Но в этом случае должен взяться следующий порт по списку. Если есть возможность воспроизвести проблему для конкретного подключения, пришлите отладочные логи...

Did you perform WCS restart? service webcallserver restart What is server version? https://forum.flashphoner.com/threads/asap-rtsp-was-working-well-20days-and-its-still-working-on-vlc.11642/#post-17530

Добрый день Какая версия сервера? Похожая проблема возникала, но была исправлена достаточно давно.

Hello We have received your server address and passwords. Please do things described above o resolve your issues https://forum.flashphoner.com/threads/asap-rtsp-was-working-well-20days-and-its-still-working-on-vlc.11642/#post-17536

What WCS version you use on server B? In your log files we can only see that stream does not play. Please, collect debug logs as described on this page and send all gathered information, including config files, to support@flashphoner.com.

HTTP would work too If you are using HTTP, make sure you open http://host:9091 ws://host:8080

You can increase sesssion refresh request limit much more, such as rtsp_refresh_requests_limit=1000 but the stream just will be interrupted less often. To play uninterrupted stream you have to change the RTSP source (IP camera device) to something more compatible with standard.

This stream has video only content. Try to add: rtp_force_synchronization=true in WCS_HOME/conf/flashphoner.properties and restart WCS server It should fix the issue. At least it works on our demo server...

Latest versions use ports 8081 and 8444 for REST API https://docs.flashphoner.com/display/WCS5EN/API+overview Try port 8081 instead 9091

Hello We raised internal request WCS-1461. We will let you know once it is checked.

Hello What is server version?

Hello If you have any troubles please make sure your WCS server is up and running https://forum.flashphoner.com/threads/troubleshooting.10768 If your server is not running, try to start it and check if it is started. https://docs.flashphoner.com/display/WCS5EN/Starting+and+stopping If you...

After /push/sound_on request you should send /push/sound_off, then /push/sound_on to play next file. As we see, your SIP sends PCMA audio. We added file WCS_HOME/conf/flash_handler_publish.sdp to make sure that RTMP stream will be published with PCMA audio track. This must help to escape...

Мы почистили файл flashphoner.properties, поскольку Вы использовали, фактически, настройки сервера по умолчанию. Проблема с настройками была в следующем: стандартный порт для Websocket соединений TCP/8080 у Вас на сервере занят, о чем сервер честно предупреждал при запуске в серверном логе...

In your case we have a transcoding from voice call stream to RTMP stream, that consumes CPU a lot. Physics can not be fooled. Try to replace this m=audio 0 RTP/AVP 8 a=rtpmap:8 PCMA/8000 with this m=audio 0 RTP/AVP 97 8 a=rtpmap:97 PCMA-WB/16000 a=rtpmap:8 PCMA/8000 in media_transponder.sdp file

It is audio codecs issue. Try to remove "speex16" from codecs list in flashphoner.properties file. We try to reproduce and will see what to do with it.

The audio only stream should be played in a browser by explicitly constraints setting in player script when stream is created, for example session.createStream({constraints:{audio:true,video:false}).play(); So you should modify Player application script...