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No. The trial license has only two limits: - time limit (30 days or up to 180 days if prolonged by request) - voice watermark in all the streams on server Your server shows it uses a valid certificate: Please try to clean coockies and site data in your browser, that's the possible reason to...

Pkease read this thread on StackOverflow and Page Visibility API documentation.

We expiriencing the issue with all the sofphones used during the test (but all of them work with OpenSIPS and Asterisk). Please clarify what softphone do you use for test? Anyway, WebRTC traffic flows correctly during our tests with your server. We opened Phone Min example page via HTTPS, as we...

Seems like Websocket connection to your server cannot be established. Please check if TCP port 8443 is not closed Seems you've commented getStream() call: That's why RTSP stream is not captured, so it cannot be republished too. You should also increase RTSP activity timeout. By default, RTSP...

Seems like you're opening the example page locally from disk file: 18:06:51,664 INFO WSServerHandler - WSS-pool-23-thread-19 Orgign: file:// Please make sure you open the page via HTTPS, for WebRTC to work. Also please note that SIP PBX is probably misconfigured. We cannot establish a...

You see SDP supported profiles, but stream metrics is not displayed. Please correct the query: { "name":"100000000000003", "published":true, "display": ["metrics"] } display is array, so do not miss square brackets. In this case, you will receive response like [ { "appKey"...

В качестве примера можно взять следующие настройки профилей (предполагаем, что публикуется H264 поток 1080p с высоким битрейтом, как рекомендовано выше, аудио Opus или AAC): profiles: # AAC to Opus 48 kHz to play in browser, video is passed as is -opus: audio: codec : opus rate ...

Good day. According to Apple policy, camera usage is forbidden in background mode, both for browsers and native applications. The only way to proceed streaming after maximizing the browser again is to restart stream publishing.

Yes. This setting is needed only to collect a report, so you can switch it to INFO or disable client logging at all enable_extended_logging=false

Good day. Please read here how to find what folder occupies the disk space. Perhaps it is server log files. Please clean them rm -rf /usr/local/FlashphonerWebCallServer/logs/* You can also clean recordings folder if you don't need recording files rm -rf...

Good day. Please check if you opening the page using HTTPS, not HTTP: https://your.domain:8444/admin/. Also, clean browser cache and cookies. If nothing helps, please provide SSH access to the server (the port 8444 should be available from outside networks) using this form.

Please check if you are using Android SDK 1.0 on Android 5-7 devices only. On Android 8 and higher, use Android SDK 1.1 (latest build 1.1.0.27). Please also check if network is good enough, this can happen due to Websocket connection interruption.

Такого быть не должно. Если подаете поток OBS H.264 + AAC, 1080p или Chrome WebRTC H.264 + Opus, 1080p, транскодинг для прямого воспроизведения по WebRTC не требуется. Основные рекомендации к толстым Full HD потокам 1080p WebRTC При тестировании следить за битрейтом по метрикам или в...

What is your stream source? Could you provide more information.

Добрый день. Для этого необходимо отключить транскодинг, который Вы включаете, указывая явно разрешение при воспроизведении. Если играть поток с тем же разрешением, с которым он опубликован, транскодинга не будет Вы можете контролировать качество канала при публикации. Обратите внимание. что...

This is probable log files. You can stop WCS systemctl stop webcallserver then clean logs folder and recording folder if there is one rm -rf /usr/local/FlashphonerWebCallServer/logs/* rm -rf /usr/local/FlashphonerWebCallServer/records/*

We also cannot access the server by SSH credentials you've sent Seems like something broken on router/NAT, please check.

We've tested RTMP republishing using your flow (capturing RTSP, then republish as RTMP) with your server. There is the following message in server log when trying to publish RTMP stream to Youtube: 03:00:30,253 ERROR RtmpPublisher - Thread-144 error creating client connection: Connection...

Unfortunately, SIP traffic cannot be identified in pcap dump. Perhaps you've started dump collection after call establishing, but it must be started before this. On your server we saw the following messages in server logs: 13:13:08,673 WARN AbstractStunSocket - STUN-UDP-pool-40-thread-1 Can...

You can detect browser and device by checking navigator.userAgent, please see example here. Then, define tha constraints depending on device, for example function publish() { ... var constraints = getConstraints(); publishStream = session.createStream({ name: streamName...