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Please clarify: - What WCS build do you use? - What player example do you use to play: HLS.JS Player minimal, VideoJS Player minimal? - Steps to reproduce a problem from publishing a stream to playing it. There is no autoplay option for HLS playback, you should just get an HLS stream URL and...

Good day. Seems like the server you've provided is no a WebCallServer instance. WCS uses is own websocket signaling, and the server doe not support it. You can play the stream via WCS using RTSP or RTMP protocol.

We're testing the fix. Perhaps it will be released next week. Please be patient.

Добрый день. В очередных логах мы видим, что для всех потоков, которые не записались, в серверном логе не было сообщений о получении ключевого фрейма и создании файла записи. При этом поток 190499-150046_6E72F232-92E0-4827-BBAD-EA62E4606F4B был остановлен через 8 секунд после начала публикации...

We reproduced the issue playing the stream from the dump provided and raised the ticket WCS-4129, We'll let you know about progress here.

Добрый день. Нужен полный отчет с логами и настройками сервера: Сбор отладочных логов при помощи скрипта report.sh. Пришлите его через эту форму.

WebSDK does not support track adding or removing. Perhaps we will support it in the next major version, but it is not a priority task and will take a long, long time to release.

Good day. You don't need to remove audio track to mute it. Use Stream.muteAudio() method, for example (Conference sample source code) $("#localAudioToggle").text("Mute A").off('click').click(function(){ if (stream.isAudioMuted()) { $(this).text("Mute A")...

We placed IDecodedPcmInterceptor implementation to a separate package com.flashphoner.pcmInterceptor, tested on JDK 12, 14, 17, 18. It seems to be working. So we updated the example in documentation: Server audio processing. Please try to build it as described.

Good day. We found a probable reason for SocketException and fixed it in build 5.2.1963. We also added some logging to detect a problem if it persists. Please update and check.

Good day. This is a known issue: a fixed 5 seconds timeout is used to check if a build is available to download. This timeout is enough in most cases, but may be not enough if there are som temporary network issues between server to update and the builds file storage. There is the ticket...

Send indication is sent by client. If client stops sending suck packet it probably stops sending a media traffic too.

Yes. According to the dump and explanation provided, a problem is between TURN server and publishing client, not between WCS and TURN сервер.

No. This is definitely a problem between publishing client and TURN server. Yes, the bottleneck may be in network settings.

Good day. Unfortunately, you've not provided any additional information to the dump: 1. TURN server address, WCS server address, client address 2. Was it publishing or playing client? We see in dump a periodic Previous segment not captured messages with TCP retransmission. It seems like a packet...

We checked publishing and playback on your server. When using a valid server domain name (we get it using reverse DNS lookup), a test stream is playing successfully in Player example in iOS Safari 17.4.1. But Safari refuses to connect to secure Websocket when connecting by IP address So you...

This is WebRTC library implementation behaviour. We can't change that.

Good day. We fixed the issue in WCS build 5.2.1960. Please update and check.

We tested WebRTC playback in iOS Safari 17.4.1 (iPhone 12 mini) on our demo server (WCS build 5.2.1917, WebSDK build 2.0.240): 1. Published a test stream in Media Devices example in Windows Chrome 123 2. Played the test stream in Player example in iOS Safari 17.4.1 The stream is playing...

Good day. Please read this article about RTSP streams load testing: How many RTSP cams can I connect to the WebRTC server?