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To get this working you have to configure 1) TURN server on port 443 with valid SSL certificates 2) HA Proxy on port 443 with valid SSL certificates for balancing between WCS ports 9091 (http) and 8080 (websocket). You can also configure WCS directly use 443 port in server.properties file...

Hello Please take a look at our firewall traversal example: https://wcs5-eu.flashphoner.com/demo2/firewall-traversal-streaming This example uses our turn server turn.flashphoner.com You can install and deploy your own TURN server that will relay all WebRTC traffic over HTTPS 443 port. You can...

Hello The WS Player was designed for iOS Safari browser only and is not tested carefully with Chrome. Currently we have two versions of WS Player 1) WS Player 1 You can see this player on our old demo page: https://wcs5-eu.flashphoner.com/demo/websocket-player-min 2) WS Player 2 This player...

We plan to implement such type of re-publishing soon. I will keep you informed over this thread.

There are two ways 1. WebRTC to RTMP from Chrome browser Example: https://wcs5-eu.flashphoner.com/demo2/webrtc-as-rtmp-republishing 2. Answer with rtmpUrl on REST hook, then it should work with any streamer. However we did not test this.

Currently we don't have API that starts pushing by external command. However we plan to implement this in the nearest future. We plan to provide REST API for re-publishing a particular stream. For example http://host:9091/RESTStream/startRePublishing { "name":"stream1", "pushName":"stream1_1"...

Проблему удалось воспроизвести и найти ее причину. Похоже, что не поддерживается метод RTMP-протокола receiveVideo. https://www.adobe.com/content/dam/Adobe/en/devnet/rtmp/pdf/rtmp_specification_1.0.pdf Попробуем добавить его обработку. После этого стримы должны начать работать. Сообщу, как...

Сейчас SWF - объект стало видно. Можно тестировать дальше.

Не совсем понятно, как плеер должен работать. Например, я публикую поток 555 с https://wcs5-eu.flashphoner.com/demo2/flash-streaming на наш сервер и здесь же забираю, чтобы его проверить: Далее открываю ваш плеер и пытаюсь подключиться: Кстати я не вижу там Flash - объекта, который бы...

Regarding load balancing architecture, In latest versions you can use RTMP re-streaming for loadbalancing. For example, if you publish a stream to WCS1-orign, you can re-stream this as RTMP to WCS2-edge. To do that you have to insert filed 'rtmpUrl' into Stream object. Example for Web SDK...

Outgoing RTSP connection will be dropped once your edge WCS server is disconnected. Your edge WCS server will close its client RTSP connection once no viewers connected to this RTSP URL. You can terminate RTSP agent via REST API using commands: Find all RTSP agents...

On publisher disconnect, WCS should terminate the published stream and all subscribers of this stream. If the publisher has lost network connection, it may take some time to terminate the connection by keep alive. It was fixed from build 2075. RTMP connections were not released in older builds...

Unfortunately we can't reproduce this issue in our environment. Our env OpenSIPs Android SDK latest iOS SDK latest WCS server build 2099 Please provide your versions of - WCS server - Android SDK (app) - Used devices If your Asterisk is a publicly available, please provide two extensions for...

Hello The developer guide is correct. Please check Web SDK home page. As you can see from this page: 1. Source code https://github.com/flashphoner/flashphoner_client/tree/wcs_api-2.0 2. Builds https://flashphoner.com/downloads/builds/flashphoner_client/wcs_api-2.0/ To get this working you have...

если 3 подключения, то идут они пачкой , а не по интервалу , каждый отдельно StreamKeepAliveEvent отправляется в отдельном внутреннем потоке исполнения thread и вызывается для всех стримов приложения (app). Т.е. при проходе запросы будут идти один за другим без каких-либо интервалов. И если у...

По-умолчанию, 403 игнорируется. Поэтому в методе connect, нужно явно задать конфигурацию для StreamKeepAliveEvent В этом топике обсуждалось как это сделать: http://forum.flashphoner.com/threads/Пароль-для-публикации-потока.10760/ Т.е. нужно явно передать такое поле в ответе на connect: ...

It is not processes. It is internal java threads. That is normal behavior. Regarding possible memory leak, we added keep alive pings for RTMP connections in build 2075. The issue was disconnected RTMP streams were not released. So if you update to the latest build there is a chance the issue...

Thanks for the report Let me resume the issue report: - Android or iOS, Phone Min Video - Call is dropped in 1 minute - WCS detects disconnect and sends BYE request to Asterisk server As I see from logs, the mobile app closes its connection by keep alive timeout because network is overload by...

The root cause of issue is your websocket connection is closed by your browser in 40-50 seconds. Try to open a page like Two Way Streaming and do connect then wait for 1 minute. If it is disconnected, it looks like our issue. If it is not disconnected, it seems your server configuration issue...

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