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We just tried to register with both accounts you've provided. The SIP server breaks the connection after first SIP REGISTER packet sent to port 5061: but when we trying to register to the port 5060, the flow goes correctly (but server returns 403 Forbidden): Unfortunately, we cannot check...

Good day. There are a various ways to check server state and stream metrics on server side. On client side, you can check a channel quality and collect browser WebRTC statistics. To detect stream playback failure, you should use custom backend which should handle /StreamStatusEvent REST hook

If upgrade to latest build does not help, please provide SSH and publishing access to the server using this form, we will test.

Please clarify how much worse? Are there some clicks, pauses and so on in mixer output stream? Please check if you set opus.encoder.bitrate parameter as we recommended. Also, check server CPU load using top for example. If CPU load is constantly high, consider to upgrade the server to more CPU...

Поскольку мы оцениваем пропускную способность локальной сети, предположим, что среда передачи полностью прозрачна, и определим минимально необходимую пропускную способность адаптера. Как правило, физические сетевые адаптеры выпускаются со следующей максимальной пропускной способностью: 100...

Эти параметры не рекомендуется менять, они выбраны по умолчанию именно для случая стриминга 4K потоков и относятся к буферизации отправляемых и принимаемых пакетов на стороне сервера. Исходя из приведенных исходных данных, пропускная способность локальной сети должна быть не ниже 10 Гбит/с на...

With SIP credentials you've provided we receive 403 Forbidden response from SIP server, but REGITER flow looks normal. Please test a call. If this still cannot be established, please provide actual SIP credentials using this form, we will check.

There is Player example, please see its code on GitHub. You can build custom player based on the code. The stream should be published on WCS server before playing it, othewise client will receive an error.

First, all the files should be converted to MP4 with the same codec (AAC), channels count (stereo), sample rate (48 kHz to prevent excessive resampling) 1. Set the following parameter in WCS configuration file flashphponer.properties mixer_idle_timeout=10000 opus.encoder.bitrate=128000 2. When...

We finally checked your server. Seems like you've set a wrong IP address in ip and ip_local settings So WCS cannot bind SIP signaling port at all. When changing IP address settings, WCS binds SIP signaling port (30001 for first SIP registration by default) and sends SIP REGISTER to SIP server...

Hello! Streaming quality is often limited by network bandwidth rather than CPU and RAM. Network bandwidth at the rate of 1 stream with a bit rate of 1 Mbps takes about 2 Mbps of channel bandwidth. For high quality broadcasts without freezes, artifacts, etc. the channel should be loaded no...

Good day. We cannot reproduce the issue in latest build 5.2.912. Please check if the following parameter is set in flashphoner.properties file h264_strict_kframe_detect=true Pleas also check: - if keyframes are sent regularly from RTMP encoder - if WebRTC stream published from browser can be...

Good day. Unfortunately, no. You should concat the required files to the one using ffmpeg for example. Yes. You can play a VOD-live stream directly from browser as described here

Please clarify the case: do you want to publish a stream to WCS from webcam, then republish this stream to a third party server, then publish the stream from third party server to Facebook using ffmpeg? If yes, the common response is yes, you can republish the stream to Facebook as is. But this...

This is a good enough way. So, you're publishing AAC stream from a file, then playing in browser via WebRTC as Opus (browsers do not support AAC playback). In this case, audio is transcoding fro AAC to Opus. You can increase the quality by Opus encoder bitrate setting in flashphoner.properties...

Good day. You probably testing the Media Devices example. The example lists all your connected devices available to a browser, to show how to select a camera and switch between them Also the example shows how to test if camera is available to browser and how to release the camera when "Test"...

Yes, this is a normal behaviour. You can keep the service running as root because the critical vulnerability which can affect root privilegies was fixed in build 5.2.780.

We tried to make a test call via your server again, with the same result: The ports 8443, 8444 seems to be opened locally on server, but are closed on router/NAT. Please provide a public instance (AWS, DO, Google Cloud, Yandex Cloud etc) and a public SIP server to test, or provide a fully...

try cd /usr/local/FlashphonerWebCallServer/bin sudo ./webcallserver start standalone It should print start log into stdout. Some JVM options can be not supported in Java 12. You will have to comment these options in wcs-core.properties to start correctly.

Hello All push RTMP streams (WebRTC as RTMP) are encoded with H.264 AAC by default. Audio encoding H.264 + Opus 48000 (WebRTC) > H.264 + AAC 48000 (RTMP) Audio + Video encoding VP8 + Opus 48000 (WebRTC) > H.264 + AAC 48000 (RTMP) You can check http://host:8081?action=stat and...