Search results

This is Chrome browser console logs to illustrate websocket signaling. The signaling is the same for Web SDK, Android SDK and iOS SDK. We raised the ticket WCS-3272 to add logging to onStreamEvent handler. Will let you know results here.

Seems like bit stream from RTSP camera is broken, or WCS cannot parse it correctly. We raised the ticket WCS-3271 to investigate this. Will let you know results here.

Good day. We cannot reproduce the issue with Samsung S10+ (Android 11, OneUI 3.1), Samsung Browser 14.2: a stream puiblished on our demo server (5.2.998, Web SDK 2.0.180) is playing in Two Way Streaming example successfully. Please try to reproduce the isssue on demo server...

When you pressing PLAY, the following happens: 1. playStream request is sent via websocket 2. notifyAudioCodec and setRemoteSDP are received from WCS 3. notifyStreamStatusEvent is received from WCS. It contains muted/unmuted status (see below) 4. playStream?.onStreamEvent is called So answer...

Добрый день Да, если Вы используете кастомный скрипт, необходимо указать полный путь к нему в настройках on_record_hook_script=/path/to/on_record_hook_script.sh Скрипт должен быть доступен на чтение и выполнение пользователю flashphoner, от которого запускаются по умолчанию последние сборки WCS

1. Start first instance with monthly license. 2. Make a snapshot or launch template based on this instance. https://docs.flashphoner.com/display/WCS52EN/AWS+load+balancer+with+auto+scale+quick+setup#AWSloadbalancerwithautoscalequicksetup-3.CreatenewLaunchConfiguration 3. Start your first...

We've done the following test with the latest MediaDevicesSwift example build: 1. Connect to our demo server, publish stream streamName 2. Tap Local Settings. In this view, switch Mute Audio on, then tap Back 3. Play the stream streamName 4. Tap Remote Settings. In this view, we see Audio...

You should wrap Session.connect() in try/catch because the connection is establishing in this method, not in Flashphoner.createSession() method. Unfortunately, we cannot fix neither symptoms nor reason if there is no consistent reproduction.

Добрый день. В сборке Android SDK 1.0.1.81 добавлена возможность определить при проигрывании, заглушено ли аудио или видео в потоке. Подробнее здесь (код идентичен тому, что используется в Android SDK 1.1, т.к. единственная разница между ними в версии WebRTC библиотеки)

The fix is on testing stage. We'll let you know about progress.

Please try Media Devices Swift example (source code on GitHub). Is the issue reproducing in this example?

Для того, чтобы появились идеи, необходимо воспроизведение. В примере 2Players проблема не воспроизводится. Следовательно, от Вас необходима минимальная модификация кода примера, при которой проблема начинает воспроизводиться. Минимальная - это значит не переписывая весь пример на React и не...

Good day. Please clarify what browser do you use? Also please check if Canvas Streaming example works in this browser. If yes, please provide your test code using this form, we will check.

Good day. You can use server statistics page. For example, to check how many WebRTC streams are publishing and playing, use the following URL curl -s 'http://localhost:8081/?action=stat&params=streams_webrtc_in,streams_webrtc_out' You'll get a response like this -----Stream Stats-----...

Please provide the player.js code which you've modified using this form, we will check

This is our demo player.html https://demo.flashphoner.com/client2/examples/demo/streaming/player/player.html This is demo player.js https://demo.flashphoner.com/client2/examples/demo/streaming/player/player.js If you have a project you have own player.js or similar file on your web server...

Добрый день. Ошибка No remote sdp available, говорит о том, что режим "Delayed Offer" не поддерживается. Это значит, что ваш PBX (Asterisk) либо MicroSIP совершает исходящий звонок по направлению MicroSIP > Asterisk > WCS > Browser, отправляет SIP INVITE запрос и в этом запросе нет SDP. Если в...

Yes. This code can be applied to player.js

Please check if this crash related to this your topic. Also, please check if the issue is reproducing in Phone Min example, Streaming-min example and so on. If not, modify example code minimally to reproduce the issue and send the code using this from.

Please change to var rtmpUrl = "rtmp://localhost:1935/live/" + "rtmp_" + streamName; Also, try to publish in WebRTC as RTMP example as we shown above.