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Good day. We added iOS Call Kit Demo Swift example using Call Kit and push notifications to receive incoming SIP calls. Note that this example works with iOS SDK build 2.6.80 and WCS build 5.2.1164. Also, it cannot be tested with our demo server because APNs credentials should be set on the...

Seems this is hardware setup issue. In traffic dump we see big packets for all the streams every 30-40 seconds Perhaps it is keyframe. If this cannot be assembled, a stream will freeze. Please check the device settings: can keyframes be sent more often (every 5 seconds for example)? In this...

To capture audio from canvas, use AudioContext. See Canvas Streaming example on GitHub: function createCanvasStream() { var canvasContext = canvas.getContext("2d"); var canvasStream = canvas.captureStream(30); mockVideoElement = document.createElement("video")...

Seems like issue is reproduced on your server after 4 hours of testing. We downloading the traffic dump and will check it.

We successfully started to collect traffic dump and restarted all the streams capturing using private IP. Will check a bitrate periodically to detect the issue.

We tried to start traffic dump collection tcpdump -npi any -B 10240 -w log.pcap tcp port 554 But seems like tcpdump is not installed, and we have no sudo rights to install and launch this. Please add user to sudoers.

Unfortunately, we cannot connect to the server from EU host using credentials you've provided: ssh: connect to host **.***.***.*** port 22: Connection timed out.

We tried to reproduce the issue on our test WCS server (build 5.2.1162), and this is not reproduced during 6,5 hours test. If you provide SSH access to the server, we try to reproduce it in your environment and collect traffic dump to analyze. But it should not be a production server because we...

Please try to exclude router if WCS and the device are under the same NAT. Use private address, not public.

Unfortunately, we cannot play the stream from traffic dump because it's filtered by source address, and there are no requesats from WCS to RTSP camera (OPTIONS etc) Now, we trying to reproduce the issue on our test server with 16 RTSP cameras you've used in your test, and the issue still is not...

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...