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Hello We need ready to use access details. If the first and last name is required, you can set a name of somebody in your team. Please note. The forum support is depersonalized. So we require ready to use access details. If you prefer personalized (priority support), please request a quote at...

Good day. We've set up sample CDN of 1 Origin and autoscaling group of Edges. Now we working on load balancing setup. Thank for your patience.

Good day. According to this recommenations based on our tests, 8 vCPU, 16 Gb RAM server is the minimum for the described case, without video transcoding. If, however, a part of WebRTC subscribers wilr use VP8 codec, there will be stream transcoding (RTSP streams are captured in H264), in this...

In logs provided we see no issues, RTMP published stream recording seems to be started correctly according to record_flash_published_streams=true{/ICODE] parameter. Then logs are interrupted, so we don't see /stopRecording query. Please collect the report again by the following way: 1. Enable...

Good day. Please set the following two parameters in flashphoner.properties file instead of obsolete rtp_activity_detecting rtp_activity_audio=false rtp_activity_video=false Client side receives STREAM_STATUS.FAILED in this case, so you should handle it and display something to the publisher...

Good day. Yes, this message can be displayed if you're trying to play WebRTC via unsecure connection. WebRTC requires secure Websocket (wss) connection to be played. If you need to use unsecure connections, consider WSPlayer or preferrably HLS.

All the recordings are stored to /usr/local/FlashphonerWebCallServer/records folder sip_record_stream parameter enables only SIP as stream recording, i.e. this call should be started by REST API query /call/startup with toStream option, for example: POST /rest-api/call/startup HTTP/1.1 HOST...

Good day. This is the most preferrable way if you can not provide direct access to RTSP stream.

And Case 3. Maybe you just need publish and play WebRTC stream? 1. Publish WebRTC https://demo.flashphoner.com/client2/examples/demo/streaming/two_way_streaming/two_way_streaming.html 2. Play WebRTC https://demo.flashphoner.com/client2/examples/demo/streaming/player/player.html This is the...

Case 2. Re-publishing to external server (your case). Are you sure you need RTMP re-publishing to external server? Because if Yes then you would need to install such an external server or CDN.

There are two different cases Case 1. WebRTC publish RTMP playback Here you don't need external server. How to test 1. Publish a stream. https://demo.flashphoner.com/client2/examples/demo/streaming/two_way_streaming/two_way_streaming.html 2. Play as RTMP in Flash player...

You can do that from CLI 1. Connect to ssh localhost ssh -p 2001 admin@localhost 2. List apps. show apps 3. Update FlashStreamingApp if you publish from OBS or live encoder update app -l "http://10.10.10.10/app" FlashStreamingApp or Update defaultApp if you publish via Wevsocket/WebRTC...

You should set up a backend server to handle /connect REST hook, username and passord will be passed to it. Please look at user authentication by domain example that can be used as basis for your custom backend setup.

Good day. Please collect a report including client debug logs as described here and send us using this link. Another option is to provide us SSH access to your server using this link, we will check stream recording in place.

Please check if target server for RTMP stream republishing is available, RTMP port ia not blocked on this server, and sofware to capture the stream (WCS, Wowza, Nimble etc) is already running

The record parameter affects stream publishing only, but not SIP calls. To record SIP call streams, you should rework backend: 1. Arrange conference on Asterisk 2. Clients makes SIP call to the conference using WCS Android SDK 3. Cilent side or backend initiates SIP as stream call from WCS to...

We just tried to reproduce the issue: 1. Published RTMP stream (because WebRTC publishing does not work due to closed media ports) to your server 2. Re-published stream as RTMP to our demo server 3. Played re-published stream on demo server So your server seems to be working. Please chaek...

Yes, in iOS stream is not available on client also.

Добрый день. Если Вы еще не покупали лицензию, нажмите кнопку Buy на этой странице и следуйте инструкциям на экране (язык можно переключить на русский). После того, как вы получите лицензионный ключ (это будет другой ключ, не триал, которым вы пользовались ранее), необходимо деактивировать на...

Good day. You can define custom video element and pass it using remoteVideo option while creating a stream to play. For example, to weak Two Way Streaming example: 1. Add video element to player div in two_way_streaming.html file ... <div class="col-sm-6"> <div class="text-center...