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Good day. Please clarify the environment: - operating system - WCS build you've running - WCS build you've updating to Please also try to install a new build aside an old one as described here. If nothing helps, provide SSH access to the instance using this private form. Please also note that in...

Good day. Usually, browser cares about it automatically while publishing WebRTC stream. Browser can start publishing from lower resolution and then raise it to requested one (but not higher). Also, browser can change publishing bitrate to fit the channel. Please read some theory in this...

Good day. Please add the following strings to setup script before WCS startup ... USERS_CONFIG=/usr/local/FlashphonerWebCallServer/conf/database.yml if grep "demo:" $USERS_CONFIG > /dev/null 2>&1; then sed -i -e "/demo:/s/active:\ true/active:\ false/" $USERS_CONFIG fi # Start WCS after...

Добрый день! Рады, что у вас все заработало! Такое поведение микшера (зависание потока, фризы и незапланированное разрушение микшера) как правило, может быть вызвано проблемами с каналами публикации. Микшер работает, пока все входящие в него потоки либо не завершатся, либо не будут удалены из...

Good day. We restored the VR example on demo server. Thank you for attention.

Please do the following: 1. Login to the server via SSH as root 2. Execute the following command to edit on_record_hook.sh script nano /usr/local/FlashphonerWebCallServer/bin/on_record_hook.sh 3. Comment out two echo strings and add the following strings chown wwwmojostreaming $FILE_NAME chgrp...

Good day. Please clarify your case: - how do you publish a stream? - how do you play a stream? For example, if you're publishing from RTMP encoder and playing as WebRTC in browser, there can be packet losses on players channel. In this case, switching to TCP transport may solve a problem...

Good day. According to you previous posts, you're publishsing stream from RTMP encoder, In this case, you should set interval for keyframes to be sent more often. For example, in OBS settings: You can also switch to TCP transport for WebRTC playback to reduce packet loss session.createStream({...

Hello Did you configure security groups? https://docs.flashphoner.com/display/WCS52EN/WCS+in+Amazon+EC2#WCSinAmazonEC2-WCSautomaticdeploymentfromAmazonAWSMarketplace Step 6

We raised the ticket WCS-3039 about this, and let you know result here. As temporary solution, use to REST queries as recommended above.

The stream shoud stop immediately after you sent deleteStream command. Please note that FCUnpublish command is not supported. You can use /stream/find REST query to find stream mediaSessinId, then /stream/terminate to terminate it.

This is beyound the scope of the support forum. But you can read how to send REST API queries from Javascript code here or try to search here

We use the forum as public bug tracker. This topic is tagged with ticket id, and we'll let you know about ticket state changing (usually, when it will be fixed) And in Web SDK too. But in iOS SDK constraints are historycally limited.

The autoplay policy in Safari requires a user action to start sound playback. Unfortunately, there is no workaround.

You can mute participants audio on client side publishStream = session.createStream({ ... }).on(STREAM_STATUS.PUBLISHING, function (stream) { stream.muteAudio(); }); publishStream.publish(); but in this case you have to unmute on client sude also. If you want a speaker to control...

You can use on_record_hook.sh script for any operations on recording files, including permissions changing, for example STREAM_NAME=$1 FILE_NAME=$2 echo $STREAM_NAME:$FILE_NAME >> /usr/local/FlashphonerWebCallServer/logs/record.log chown wwwuser $FILE_NAME

Good day. Please clarify your case: 1. Do you publish a stream to WCS from RTMP encoder rtmp://wcs:1935/live/stream? 2. Or do you republish a stream from WCS server to a third party RTMP server? In both cases, you cannot publish a new stream with the same name until previous stream is stopped...

Скорее всего, WebRTC библиотека (которая используется под капотом Android SDK) в SDP для сервера все таки показывает видео дорожку в этом случае, хоть и не отправляет медиа данные. По битрейту видео: если в потоке нет видео, значение метрики VIDEO_RATE будет равно 0. Кроме того, понадобится...

В таком случае, попробуйте протестировать по нашей методике, с публикацией потоков с помощью стандартного примера "Two Way Video Chat" Если проблема не воспроизводится, модифицируйте код примера таким образом, чтобы проблема воспроизводилась и пришлите для анализа. Форма для отправки

Works in latest 5.2.878: stream is added to mixer as video-only. If it does not work, please describe in which case.