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Здравствуйте. 1. Работает ли пример Two Way Streaming - на вашем сервере - на демо сервере https://demo.flashphoner.com/client2/examples/demo/streaming/two_way_streaming/two_way_streaming.html 2. Воспроизводится ли ваш URL в плеере VLC? rtsp://333.333.333.333:45183/live/ch00_0 3. Пришлите...

Добрый день. Для работы WebRTC необходим домен, привязанный к серверу, и актуальный SSL сертификат. Пожалуйста, зарегистируйте домен, приобретите сертификат и импортируйте его, как описано здесь: Websocket SSL.

We reproduced the issue on our test server. Raised the ticket WCS-3866 to investigate and fix it. Please do not shut down the accounts, we'll use them for reproducing and testing.

Please provide for us to test and reproduce the issue: - two SIP accounts (caller and callee) - the detailed description of the operation flow using two Phone Video examples Please use this form to send.

We test a call from Phone Video example to Phone Video in Chrome desktop browser, and SIP as RTMP call (with both audio and video enabled) to Phone Video example in Chrome desktop browser. We can't reproduce the issue with build 5.2.1691. Please clarify what script do you use and what do you...

Good day. The issue was fixed in build 5.2.1691, now video calls should work correctly with generate_av_for_ua=all. Please update and check.

Please enable the old stream injector implementation by the following parameter (WCS restart required) use_new_injector=false Then use /stream/inject/startup to inject VOD stream (not inject2!). We also raised the ticket WCS-3863 to implement VOD inject function in new stream injector.

Hello 5.2.1690 uses new injector implementation by default. Try use_new_injector=false Please provide related error logs. We will try to reproduce issue in our environment. You can provide logs right here or via the private form Collect error report. It will help us to fix the issue.

Добрый день. Миграция на Swift Package Manager затруднена, поскольку в нашем iOS SDK есть зависимость, которая поддерживает только CocoaPods. Есть PR, но разрабочики его игнорируют.

Good day. There are no ready to use React components. You can use React integration examples (source code: JS, TS) to integrate Flashphoner WebSDK to React application and create your own component. Please use SIP Phone example (source code) as a base code for the component.

We still working on it. We'll let you know when issue will be fixed.

Good day. Please provide an SSH access to the server and SSL certificate files archive using this form, we will check.

Good day. Google claims this issue is not mandatory to fix. So you can still use the latest Android SDK 1.1 build. We raised the ticket WCS-3843 to update WebRTC library to actual version. But this will definitely lead to drop Android 8-11 support, depending on minimal libwebrtc API level...

We reproduced the issue and working on it in the ticket WCS-3837. Let you know about progress here. A quick and dirty workaround is to disable A/V generator, but in this case only video to video and audio to audio calls will work.

iOS SDK uses WebRTC library without modifications (just built from sources). The library established WebRTC connection and sends media packets correctly, but audio packets contains empty media payload as we checked above. So this seems to be iOS issue, not Flashphoner iOS SDK. This is the common...

Good day. We checked the report. According to client logs and traffic dump, an application successfully sends media packets (we can play the stream from traffic dump), but audio packets are sent with empty payload (0 bitrate). So you should remove the following lines from flashphoner.properties...

11:09:57,448 ERROR FileRecordRotation - FScheduling-pool-1-thread-1 Problem to move file /usr/local/FlashphonerWebCallServer/records/2306024Eljumbp-shareD1685671629464-377c_{startTimeMillis}_{endTimeMillis}.webm to...

Hello If you are using rooms API, we assume that you may use multiple recorder. Because this recorder can record video tracks in sync with audio. That's why we supposed the multiple recorder use. According your notice it seems you don't user multiple recorder. If so, all streams in the room...

SIP calls recording is supported only for SIP as stream or SIP as RTMP functions. If you're using WebRTC-SIP gateway, the only way to record is to setup call recording at SIP PBX side. There is also the ticket WCS-3723 to fix SIP calls audio recording, but no ETA for this ticket yet due to more...

Good day. Please read this step-by-step guide to configure a simple CDN.