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Статус NOT_ENOUGH_BANDWIDTH отправляется тогда, когда потери пакетов на канале зрителя оцениваются как более 5% Уменьшение битрейта уменьшит потери на канале. Уверичение порога уменьшит количество отправленных NOT_ENOUGH_BANDWIDTH webrtc_cc2_bitrate_overuse_event_threshold=0.05...

To migrate to another instance you can use following procedure: 1. Create a snapshot. 2. Create a new instance using the snapshot. 3. Sign in using SSH and change ip= and ip_local= settings in /usr/local/FlashphonerWebCallServer/conf/flashphoner.properties 4. Restart WCS. service webcallserver...

Проблемы при воспроизведении WebRTC потока в Safari объясняются просто, Safari не любит резких изменений битрейта. причем это касается и Safari на Mac. В этом случае рекомендуем обновить сервер до последних сборок, начиная с 5.1.3753 восстановлено управление битрейтом через constraints. и зажать...

Hello. To play this file as VOD you should use WCS iOS SDK, for example (iOS Player application): - (FPWCSApi2Stream *)playStream { FPWCSApi2Session *session = [FPWCSApi2 getSessions][0]; FPWCSApi2StreamOptions *options = [[FPWCSApi2StreamOptions alloc] init]; options.name =...

Hello. Try to change your instance type as described in AWS docs.

Hello. The latest WCS builds support image rotation for RTMP streaming, please read this doc.

Hello. Build 5.1.3740 is stable. There was no changes for WebRTC over TCP or built-in TURN in latest builds, but there was some optimizations for more stable server work in CDN under high load. So we recommend to upgrade if your WCS serves many streams or if you're using CDN.

Hello What is server version? 1. Try to update to the latest one 3762 https://docs.flashphoner.com/display/WCS5EN/Release+notes 2. If update does not help, try to enable TCP https://docs.flashphoner.com/display/WCS5EN/Publishing+and+playing+stream+via+WebRTC+over+TCP 3. Finally if options 1 and...

Yes you can publish to any RTMP urls either with the same hostname or different. Please check out docs for more information: https://docs.flashphoner.com/display/WCS5EN/To+another+RTMP+server

Hello. You cannot publish one stream to two servers using WebSDK rtmpUrl property. But you can easely push one stream to two RTMP servers using REST query /push/startup. For example, you publish WebRTC stream named test using Two Way Streaming example. Then, you make two REST queries...

This is list of requirements for each type of license: Trial Initial network access for activation (HTTPS) Permanent network access to pool.ntp.org (NTP) Monthly subscription Permanent network access to https://my.flashphoner.com (HTTPS) Standalone license Initial network access for...

You use tv_app for RTMP broadcasting. So, to play your stream as WebRTC you should add another application, for example add app Play_app play_app "http://your_host/rest" add app-rest-method -a play_app and pass its key Flashphoner.createSession({urlServer: url,appKey: 'play_app'}) Or, you can...

Hello. Please clarify: do you use one of our proxies (proxy1 or proxy2) as described in docs, or some other proxy? proxy1.flashphoner.com and proxy2.flashphoner.com does not require authentication. But, if you use your own proxy, you should set up it as transparent with no authentication.

Hello. We have published RTMP stream to your server ffmpeg --re -y -rtbufsize 1k -i file.mp4 -preset ultrafast -vcodec h264 -acodec aac -ar 48000 -strict -2 -f flv -b:v 1M rtmp://your_host:1935/tv_app/flashphoner_test and played with Embed Player...

You probably add your applications like this: add app Live live "http://bro4bet.live/rest" add app-rest-method -a live That was right. But if you broadcast RTMP streams to WCS server, you should aslo do this update app -m com.flashphoner.server.client.handler.wcs4.FlashStreamingHandler -c...

Hello. If you publish RTMP stream using your server application, you should set corresponding handlers for direct calls and callbacks. Default handlers (WCS4Handler and WCS4CallbackHandler) works with WebRTC streams, not RTMP. For example: update app -m...

Hello. Android SDK implements WebRTC streaming. WebRTC does not allow to change picture ratio or rotate picture on the fly. Orientation can be changed for RTMP streams only, see doc here.

Hello. Please check if telephone_event codec is in codecs list: codecs=...,telephone-event,... and is not excluded from SIP codecs list (telephone_event should not be in the following list) codecs_exclude_sip=mpeg4-generic,flv,mpv,opus,ulaw,h264,g722,g729 Also, prepare report as described here...

Hello. To play the file via VOD, you should place it to /usr/local/FlashphonerWebCallServer/media folder. On client side, file should be requested as vod://stream-7608750A-3BAE-4E0D-B84E-20910C3D1584-iiabkki005tad8omrrgr121slr.mp4 No file path should be included.

Hello. Please prepare report as described here and send to support@flashphoner.com. We will check.