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Добрый день. В наших тестах с MacOS Safari 15.3 проблема не воспроизводится в последней сборке WebSDK 2.0.217 в примере Screen Sharing с бандлом по умолчанию (flashphoner.js) или с WebRTC only бандлом (flashphoner-webrtc-only.js). Пожалуйста, обновите WebSDK и проверьте, воспроизводится ли...

We checked the logs. Seems like WCS starts to receive incomplete video frames from all the RTSP streams at the same moment. So please do the following: 1. Choose one RTSP stream on which the problem is reproducing. 2. Start to collect traffic dump on server tcpdump -npi any -B 10240 -w log.pcap...

We'll check it in ticket WCS-3465. The Chrome bug is already marked as fixed: https://bugs.chromium.org/p/chromium/issues/detail?id=1313851#c7. Perhaps it will be released soon. We'll fix orientation value at our side, and the fix will affect a new recordings. For old recordings, there are the...

If you cannot provide Session Progress to be sent to all the callers, use the CALL_STATUS.RING to play a ringtone locally on client: var outCall = session.createCall({ callee: $("#callee").val(), visibleName: $("#sipLogin").val(), ... }).on(CALL_STATUS.RING...

Unfortunately, we cannot reproduce the issue on our test server. So please collect a debug logs for the problem stream enable_extended_logging=true client_log_level=debug then collect a report using report.sh script and send using this form. If report archive is more than 28 M size, please...

Good day. This is a known Chrome bug https://bugs.chromium.org/p/chromium/issues/detail?id=1313851&q=mp4&can=2. Perhaps Chrome develepers should fix it soon. There is also the ticket WCS-3465 to check MP4 recording metadata. Use ffmpeg postprocessing ffmpeg -i stream.mp4 stream-fixed with...

Thank you for information. Don't be sorry: we'll investigate at our side why rotation is set in recordings (normally should not because WebRTC orientation extension is disabled by default).

Seeme like the camera stops sending media traffic. You can enable video RTP activity control rtp_activity_video=true rtp_activity_timeout=60 In this case, stream shpuld stop if no media traffic form the camera in 60 seconds. Then, it depends on how do you capture RTSP streams. If you start...

First, use 8 vCPU (m5.2xlarge for example). Do not use shared CPU because this may affect performance. Also, do not allocate all the server RAM for Java heap memory: a much data are stored in native (system) memory during transcoding. For 16 G RAM, the settings should be -Xms8G -Xmx16G iperf...

Looks like codec issue. Try to test Media Devices example: 1. Publish stream Media Devices https://demo.flashphoner.com/client2/examples/demo/streaming/media_devices_manager/media_device_manager.html 2. Play stream on the device where you got green video frame Player...

Hello If you provide RTSP url we will setup 2 hours long test to reproduce freeze. Report form

Good day. Plerase clarify, what server hardware configuration do you use and how much memory you allocate for Java heap? Mixer decodes all incoming streams and encodes outgoing stream, so it requires a lot of CPU. We recommend at least 2 CPU cores per one mixer plus at least 2 CPU cores for...

Yes, 183 Session Progress SDP message instructs the caller side to play a media stream according to SDP. So Asterisk should send this message and then RTP media data to the caller. Please check the Asterisk configuration.

Good day. This seems like Chrome bug affecting MP4 metadata parsing, because Firefox, Safari and VLC plays MP4 recording files normally. But we created a ticket WCS-3465 to check if metadata we writing conforms MP4 container specification. If yes, this is definitely a Chrome bug. As workaround...

Good day. We implemented stripCodecs option in iOS SDK (ticket WCS-3416) in build 2.6.78. Please see Media Devices Swift example description for detalis: iOS Media Devices Swift. We still working on multiple streams recording to MKV container in ticket WCS-3419. Will let you know when the...

We checked the dump and tested calls via your Asterisk using SIP accounts provided. Some calls in the dump contains 183 Session Progress SDP packet sent from Asterisk to caller (but not all the calls): In our test with SIP accounts provided, there is no such packet, both in calls from WCS to...

Good day. Please provide a link to RTSP stream using this private form.

Hello Quick solution: chown -R flashphoner:flashphoner /usr/local/FlashphonerWebCallServer/client2 Step-by-step: 1. Backup folder /usr/local/FlashphonerWebCallServer/client2 This folder is old Web SDK content 2. Remove folder client2 and re-create empty folder client2 3. Download and...

Hello Password should be MD5() So if password is admin, then MD5('admin') = 21232f297a57a5a743894a0e4a801fc3 admin: {username: admin, password: 21232f297a57a5a743894a0e4a801fc3, active: true}

Hello Regarding issue WCS-3463. Please check out your /etc/hosts config If your /etc/hosts looks like 127.0.0.1 myfqdn.com Then it won't work because domain must be resolved to public IP address. Therefore we marked issue as "Can't reproduce". Please provide more information how to...