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Try to check hostname It should be resolved to IP address /etc/hosts 127.0.0.1 myhost Should be something simple. if you provide SSH access to support@flashphoner.com, we will check

Hello. Please update to 5.2.28 build. In this build, {startTime} record name template parameter will be set to recording start timestamp. Also this build supports custom record name definition via REST API, see this doc for details.

Hello Please check if TCP media ports in range media_port_from:media_port_to is opened in ASW instance security rules.

Hello Restarts do not affect any license functionality. SIP functions means functions where SIP protocol is used. The SIP protocol is used in integration with external SIP servers, PBX, telephony services like Twilio, etc. You cal find all SIP functions in the docs: 1. SIP integration...

Internal ticket WCS-1850 has been created for the issue. Will inform in the thread when there is an update.

Yes, it's correct. Frontend and WCS can be placed to different servers. To avoid cross origin problems those instances should be in same domain.

Добрый день. В разделе документации "Общие сведения о функциях сервера" приведены таблицы поддерживаемых функций стриминга и WebRTC-SIP шлюза. В описании каждой функции в разделах "Функции потокового видео" и "Функции WebRTC-SIP шлюза" приведены таблицы поддерживаемых платформ и основных...

Hello. Phone-ui is JavaScript web application, it is downloaded from hosting server then executed in browser on clients' device. So requirements to hosting server are the same as requirements to usual web hosting for 100+ mutual connections.

Hello Muhammad. Please update to latest WCS build from this page. If the issue still persists, prepare a report as described here and send to support@flashphoner.com with detailed description of your case including SIP accounts to test. We will check.

Добрый день. Мы работаем над этим. Исправлений можно ожидать в версии 5.2 после выпуска некоторых важных возможностей в части CDN и транскодинга.

Добрый день. Для идентификации пользователей можно использовать REST hooks. Запросы на публикацию и просмотр потока направляются на бэкенд-сервер, где можно собирать данные из этих запросов. Пример реализации при помощи REST hook авторизации пользователей по домену с пошаговым разбором скрипта...

Hello. certbot-auto script still available in WCS 5.2. You can use it as described here.

WebRTC browser does it for you. When bitrate is limited, browsers adjusts stream publishsing resolution if bandwidth is not enough. A bad thing is that some browsers (Safari for example) may freeze when stream parameters change. So you should try to detect most stable publishing bitrate for your...

You have a very poor channel to publish. Please test your channel bandwith with iperf3 via UDP iperf3.exe -c yourserver -p 5201 -u The result will show you maximum bitrate you can publish without packets losses. So you should set up your local network (router etc), or relocate your server to get...

Hello. Is the issue reproduced with latest WCS build (5.2.25) with generate_av_for_ua=all setting?

Hello. We've checked your recording. It seems like packets are lost periodically. This confirms recommendation to either relocate your instance closer to publisher or to publish stream from US to US. You also can use WebRTC over TCP to publish/play a stream, it may help to escape packet loss but...

Hello. Please clarify the following: 1. What is your stream resolution? 2. What bitrate is shown in chrome://webrtc-internals, bweforvideo section, googTransmitBitrate parameter when your stream becomes poor? Attach screenshot if possible. 3. What is displayed in Stats graphs for bweforvideo...

Hello This means your H264 stream goes with payload type (pt) 119, but RTMP pull agent payload type for H264 is 95 by default. To fix it create file rtmp_agent.sdp with the following content v=0 o=- 1988962254 1988962254 IN IP4 0.0.0.0 c=IN IP4 0.0.0.0 t=0 0 a=sdplang:en m=video 0 RTP/AVP 119...

Hello Try to setup bitrate settings in WCS_HOME/conf/flashphoner.properties webrtc_cc_min_bitrate=1000000 webrtc_cc_max_bitrate=1500000

Hello The investigation will take a time and there is a chance it will be fixed. However to manage streams more efficiently you can use direct and raw Websocket API. https://docs.flashphoner.com/display/WCS52EN/Raw+WebSocket+API So you can manage streams using raw browser getUserMedia() WebRTC...