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Also, when we tried to register on your PBX from our test server, registration fails due to Certificate unknown error. So please check if you've done the steps described here: SIP calls via SIP PBX server with self-signed SSL certificate

Unfortunately we have not received your report. Please modify iOS Two Way Streaming example application code minimally to reproduce the issue and send using this form

Good day. No. The user action must be performed to unmute the audio when autoplaying. If audio playing volume changing after publishing a stream fro the same page. please do this two steps: Audio playback tuning in iOS Safari

Please check /usr/local/FlashphonerWebCallServer/bin/setenv.sh file. Seems like LANG is set explicilty in the file. If not, please provide SSH access to the server using this form

We checked SIP registration from your server. Seems like your SIP PBX is unreachable from your WCS server Please check your NAT/router or PBX firewall setup.

Good day. This is a known iOS Safari bug fixed in 15.2. Please read the details here (p. 17)

We checked the report. Unfortunately, the test stream (streamtest we suppose) is published from Chrome 94 on Windows 10, but you mentioned the problem is in iOS SDK application. Also, the server where report collected is the Origin, but test stream seems to be playing from some Edge server, so...

We reproduced the issue and already testing the fix. Please be patient.

Please check if WCS build, configuration and JDK are the same on both servers. If not, make the problem server setup identical to a good one. If this does not help, collect a report from the problem server as decsribed here using report.sh script, and send using this form.

Please try to reproduce the issue in Media Devices example: 1. Choose 640x480 from resolution picker in iOS example or set width to 640, height to 480 in Android example. Please note that width should be set to 640 and height to 480, see code on GitHub. 2. Publish a stream 3. Check stream...

Please read here about receiving an error info: WebSDK error handling Also, please red this article What kind of server do you need to run a thousand WebRTC streams? about tuning and testing WCS for a huge number of subscribers. See the "Preparing for testing" chapter for a useful tweaks for...

Good day. You should set the same resolution as original stream, i.e. 640x480, not 480x640.

Please reproduce the issue and collect a report on server side as described here using report.sh script. Also collect traffic dump (the collection must be started before stream publishing). Send the treport using this form.

Проверили настройки сервера и логи. По логам не видно каких-либо проблем. Проблемы, которые вы описываете, похожи на нехватку битрейта (скорости канала) между пользователем и сервером. Рекомендации: 1. Выставить минимальное разрешение видео при публикации. Например, если ваше приложение...

Пришлите SSH доступ к серверу через эту форму. Проверим настройки сервера. Также опишите пожалуйста как технически работает ваше приложение: один ко многим, многие ко многим, вебинар, трансляция, и т. д.

Hello You can manually accept not trusted certificate in browser if you open page directly: https://ip:8443 Once you accepted server's certificates, you will be able to connect by IP. wss://ip:8443 Example:

Good day. When internet connection is lost, the stream will be stopped. So, recording will be stopped too. And in this case you cannot get a link because websocket signaling session is closed too. Moreover, if you are using a static template like {streamName}, the previous recording will be...

Yes, we received this URL via email form and testing this stream. The issue is in stream media packets parsing, so please leave it available. We'll let you know when finish testing.

We checked your server. Seems like streaming is working at the time of test: But you're using the default Java heap settings (1024 M). This may be not enough in production environment. Please set Java heap to 1/2 servers RAM (16 Gb in your case) in the wcs-core.properries file: -Xmx16g -Xms16g...

Please clarify: dou you use a WebRTC-SIP call, or just WebRTC two way video chat? For WebRTC-SIP calls to receive in background, please wait for WCS-3393 implementation. For WebRTC two way video chat, please check if there is audio/video traffic between two sides (you can test it in WebSDK Media...