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Good day. There is the parameter to enable auto start HLS cut for all the published streams (not RTMP only) hls_auto_start=true But we do not recommend to use it because this can impact server perfomance. Please use REST API to start HLS cut for certain stream published, for example POST...

Good day. Those are probably different users behind the same NAT. Every client can use grey (private) IP only, for example 192.168.0.100, and internet service provider can use one white (public) IP for all those users. In this case, only source port will differ for such clients.

Please read WebRTC protocol specification here. We have no library for python client, but there is low level Websocket API. You can try to integrate it with this python WebRTC library.

Добрый день. Вы можете получить список доступных браузеру устройств ввода и выбрать нужное устройство из списка Flashphoner.getMediaDevices(null, true, MEDIA_DEVICE_KIND.INPUT).then(function (list) { list.audio.forEach(function (device) { ... }); ... }).catch(function (error) {...

Good day. In latest builds, there were some fixes in DTLS and ICE procedures in addition to IPv6 support. So please update to the build 5.2.673 and check.

Hello This is docs https://docs.flashphoner.com/pages/viewpage.action?pageId=9242392

We have already checked your server, please see this post. Seems like media ports (31003-32000/udp) are blocked on your NAT/router or not forwarded to the server. Please check your NAT/router setup. Also, we recommend to set external IP to ip parameter ip = your_external_ip and to add the...

Good day. Please see Media Devices example (and source code on GitHub). You can use Flashphoner.getMediaDevices() method to get all the input devices visible to the system, then use MediaDeviceList.getVideoList() to list available cameras: mCameraSpinner = (LabelledSpinner)...

By default, HLS is transcoded to resolution 640x480. Please set the following parameters in flashphoner.properties to disable it hls_player_width=0 hls_player_height=0 Note that audio still will be transcoded to AAC 48 kHz if stream is published using PCMA, PCMU audio codecs or AAC with...

alert() function is JavaScript function provided by browser. Probably the browser can not display a message on another window, especially if it is another application window. So this issue is out of WCS WebSDK scope. You can look for some third party libraries which allows to customize...

Добрый день. Проблема исправлена в сборке 5.2.673. Сборка включает также фикс на стороне WebSDK 0.5.28.2753.139

Hello, WCS-2759 has been fixed in WCS v. 5.2.673 (also contains Web SDK fix - v. 0.5.28.2753.139).

Hello, Reproduced issues with these settings encode_record_name=true,HEX exclude_record_name_characters=:.,/ Created tickets WCS-2773 (for hexadecimal encoding) and WCS-2774 (for exclude_record_name_characters) for the issues - will inform when there is an update. Each setting should be added...

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...