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Please test this in Chrome browser. Then share Developer Tools - Console output. It seems you don't have SSL certificate on WCS server. Your wss address should be: wss://yourdomain.com:8443 And you have to import valid (green) SSL certificate using Dashboard / Security / Certificates Once your...

You can also read developer guide to get code explanation https://flashphoner.com/docs/wcs5/wcs_docs/html/en/wcs-developer-guide-2/index.html?two-way_streaming_web_sdk.htm

There are several steps: 1. Test default streaming example: https://serverip:8888/demo2/two-way-streaming Make sure it works for you. 2. Copy folder /usr/local/FlashphonerWebCallServer/client2 to your web hosting. And test...

This is example of SIP details. We have OpenSIPs server configured. And pool of SIP accounts: 10000, 10001, etc. You should have your own SIP account.

Do you have a SIP account? You need SIP details from third party provider or from a third party PBX server. Generally SIP account includes: - login - authentication name - password - domain - outbound proxy - port (5060 by default) Do you have such details? If not, you need to get the details...

Для публикуемого потока статус FAILED не приходит. Он приходит для тех подписчиков, которые этот публикуемый поток смотрят. Можно провести простой тест. 1. Опубликовать поток в Two Way Streaming https://wcs5-eu.flashphoner.com/demo2/two-way-streaming 2. Закрыть браузер Chrome. На REST придет...

You can fix this by two ways 1. Update to the latest available build 2358 service webcallserver update There is a chance it is fixed in the latest builds. 2. Use REST API method You can terminate particular RTSP stream and avoid WCS server restart. /rest-api/rtsp/terminate {...

Don't use Music and Mute controls. Keep these controls disabled. I have checked with audio only. It works for me. Please share your 1. Server version. You can see in dashboard. 2. Server config /usr/local/FlashphonerWebCallServer/conf/flashphoner.properties You can also test with our demo...

Test description 1. We use SIP as RTMP 2 example https://wcs5-eu.flashphoner.com/client2/examples/demo/sip/sip-as-rtmp-2/sip-as-rtmp-2.html 2. We use Bria 4 SIP softphone and OpenSIPs as a signaling server. Therefore we call from WCS to Bria and then forward stream to RTMP server. Test steps 1...

This is how it works on our demo server. https://wcs5-eu.flashphoner.com/client2/examples/demo/sip/sip-as-rtmp-2/sip-as-rtmp-2.html Please check again. We updated server to the latest build 2358.

As I know, our iOS SDK does not support this feature. I will be able to provide more accurate information tomorrow.

This means video re-scaling. Example, 1. Publish video 640x480 (4:3) 2. Play video 400x225 (16:9) If resample_video=true, WCS will rescale video from 640x480 to 400x225 and it will be flattened vertically. If resample_video=false, WCS will cut video to 400x225, and part of video will be lost...

We have two REST API versions, 1.0 and 2.0 This example using version 1.0 and it is deprecated. Please use REST API 2.0. You can find docs here: https://flashphoner.com/docs/wcs5/wcs_docs/html/en/wcs-rest-api/index.html You can test this API using Advanced REST Console...

There is a timeout, about 1 minute. If no new subscribers, RTSP stream will be terminated. You can control this using REST API: Request POST /rest-api/rtsp/find_all HTTP/1.1 Host: 192.168.1.101:9091 Content-Length: 0 Content-Type: application/json Response HTTP/1.1 200 OK Server...

Hello, All licensing issues can be solved via helpdesk@flashphoner.com Please send the request above there. Please do not share license number in public domain.

Логи проанализировали. По логам и указанной сессии видно, что произошел дисконнект во время публикации потока. 1. Пользователь приконнектился, начал публиковать поток. 2. Пользователь потерял TCP/Websocket соединение с сервером. В результате, публикуемый поток был остановлен. Потеря соединения...

How do you test DTMF? When you call, for example to airport, the automatic voice says 'Press 1 to ...'. It is IVR (voice menu) and it of course should work on SIP side. For SIP as RTMP case, DTMF works over REST API 1. You can setup a call: https://host:8888/rest-api/call/startup { ... } 2...

Could you please zip and send /usr/local/FlashphonerWebCallServer/conf /usr/local/FlashphonerWebCallServer/logs/server_logs/flashphoner.log /usr/local/FlashphonerWebCallServer/logs/flashphoner_manager.log to logs@flashphoner.com We will check.

We have tested with WebView and it works for us. Make sure, you open URL over https:// in the WebView It won't work if you test with http:// because WebRTC requires https page to get access to the Webcam. So, make sure you have imported SSL certificate and it works over https and wss. Example...

The issue should be fixed in 2354 server build. Our QA is checking this now. You can also test latest build with your code: service webcallserver update 2354