Search results

Здравствуйте. Настройка кодеков По-умолчанию включены все: codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv Чтобы исключить все голосовые и видео кодеки кроме G.711 (alaw, ulaw), нужно перечислить их так...

@stone We tested the sream URL you've sent. Unfortunately, it cannot be played both with WCS and VLC with connection timed out message (see VLC screenshot below): Please check if your stream can be played in VLC. Also the stream must be available 24/7 to test it.

Please update to the latest WCS build 5.2.1725. We tested redirection with this build, it should work. If not, please provide RTSP URL to test using this form.

Good day. We fixed the issue in build 5.2.1719: VOD injection is available when using use_new_injector=true and /stream/inject2/startup: Injecting a VOD stream from file

Добрый день. Начиная со сборки iOS SDK 2.6.114, фреймворки, входящие в состав iOS SDK, доступны в виде пакетов для Swift Package manager: FPWCSApi2 - фреймворк для ObjectiveC приложений (только для обратной совместимости, в промышленной эксплуатации рекомендуется использовать Swift)...

Still working on fix. Please keep a test configuration. We'll let you know when fix the issue.

Проверили Ваш сервер. По всей видимости, WCS был некорректно установлен: отстутсвуют символьный ссылки на необходимые модули в каталоге /usr/local/FlashphonerWebCallServer/lib/so/. На скриншоте слева Ваш сервер, справа - работающий инстанс с корректной установкой При попытке опубликовать или...

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...