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Good day. You can update WCS to latest version from this page by downloading and installing new version over previous one as described here

Good day. Please clarify: 1) How do you streaming to WCS: stream is published to WCS with WebRTC/RTMP, or stream is pulled from RTSP/RTMP source? What is the stream parameters (resolution, bitrate)? 2) What browsers on what platforms you are using for stream playback?

Good day. WCS does not support Apache Flink. This product is for distribute computing, not for media streaming case. You can use streaming video CDN functions to distribute server load.

You can try Media Devices demo and test audio-only stream. https://demo.flashphoner.com/client2/examples/demo/streaming/media_devices_manager/media_device_manager.html As you can see you can uncheck "Video" both for publisher (left) and for player (right) to have audio only stream test. Latency...

Hello We raised internal ticket WCS-2006 Will inform once we have any progress on this issue.

Good day. Seems like your developer missed point 3 on step Preparing examples for building, it should be done before building examples with Gradle or Android Studio. Developer should check if he did all this step point by point as described.

Good day. Please prepare a report as described here and send it to support@flashphoner.com, we will check. Also, check your PBX SIP signaling logs for BYE message from WCS.

Good day. We cannot reproduce the issue as you describe neither modifying Media Devices example, nor Two Way Streaming (this is simpler to modify because you do not need to set deviceId in constraints) using WebSDK 0.5.28.2753-075cc8b30a361f74b3be6b5f293c528b30e2a5c1 (0.5.28.2753.90) on WCS...

Добрый день. Мы добавили в constraints по умолчанию audio: true для Chrome 74 и выше для публикации системного звука при захвате экрана без расширения в сборке WebSDK 0.5.28.2753.90

Good day. If audio devices access should not be requested while choosing a camera, getMediaDevices() function may be called with explicit constraints setting Flashphoner.getMediaDevices(null, true, null, {video: true, audio: false}).then(function (list) { ...list.video.forEach(function (device)...

Good day. We can't reproduce your issue with WebSDK build 0.5.28.2753.87 (0.5.28.2753-0e77b5a453d667d7201893d939001e904d171d3a) on WCS build 5.2.115 using Media Device example in the following browsers: Chrome 74.0.3729.131 Firefox 66.0.3 MacOS Safari 12.1 (14607.1.40.1.4) iOS Safari 12.1 (iOS...

Good day. Your developer should download and build Android SDK 1.0 samples as described on this page, if you plan to support Android before 9. Android SDK 1.1 (supports Android 9 and newer) samples instruction is on this page.

Good day. Please check an output of top command top String with "java" shows how CPU is loaded by WCS. Note that you have 8 CPU cores, so 100% in top means that only one core is fully loaded. Also, your server log is full of messages Codec not found; pt - 119; current pt - 95 This may affect...

Good day. Please try to exclude codecs unused as described here (point 4). Also, you can try to switch to TCP transport with the following setting sip_force_tcp=true if this is supported by SIP side. If problem still persists please prepare a report as described here and send to...

Добрый день. Да, такая корректировка запланирована, о результатах сообщим здесь.

Good day. We have fixed this issue in WebSDK build 0.5.28.2753.87, please update WebSDK to this build and check. When audio switched off in constraints constraints: {video:true, audio:false} browser does not request microphone permission, so video publishing works while normal phone call.

Good day. We have fixed this issue in WebSDK build 0.5.28.2753.87, please update WebSDK to this build and check. When audio switched off in constraints constraints: {video:true, audio:false} browser does not request microphone permission, so video publishing works while normal phone call.

Are you logged as demo user? Try log in as admin:admin default.

-Xmx8g for maximum heap memory pool value and -Xms8g for initial memory allocation

Because this RTMP server is available from our server: PING host_ip (host_ip) 56(84) bytes of data. 64 bytes from host_ip: icmp_seq=1 ttl=55 time=175 ms 64 bytes from host_ip: icmp_seq=2 ttl=55 time=175 ms 64 bytes from host_ip: icmp_seq=3 ttl=55 time=175 ms ... Please ping your RTMP server...