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As we see, your server still have local ip set as external: Please correct the settings as we recommended above

Добрый день. Насколько видно из скриншотов, проблема не в микшере, а в публикации потока. Поэтому проведите, пожалуйста, тесты без микшера, с одним потоком: Проверьте публикацию в примере Media Devices, он покажет, высылает ли браузер пакеты. Если не высылает, проблема, вероятно, на стороне...

We checked your server. Please set ip parameter to external IP address, in this case streaming will work: ip=110.xxx.xxx.xxx ip_local=192.168.xxx.xxx (we hide some octets for your security)

Please provide SSH access to your server instance using this link, we will check.

Good day. Yes, Flashphoner is SFU server in this terms. And no, it cannot be used with Unity render streaming package unless it will implement WCS Websoket API for signaling. Seems like Unity streaming client is compatible with its own server implementation only.

Good day. If your server is in LAN behind the NAT, the media traffic ports range (31001-32000/UDP and TCP by default) on the server must be accesible from external network. Please check your router configuration to set up media port forwarding to WCS server. Please read details here.

In the case of a 0.5 Mbps channel bandwidth only 360p stream can be played with acceptable quality. In this case, the upper limit of the bitrate must be reduced for this resolution to 400 kbps, and the lower to 100 kbps. For WebRTC subscribers, we have a channel quality detector. Focusing on...

Good day. As seen from you video sample, you have a big packet losses while playing RTMP stream. Then, you reconnect to play WebRTC, and there are no packet losses. May be it was channel issue? We tried to reproduce the issue with the latest WCS build, and can not reproduce. Please update to...

Good day. Please try to use wget as described in docs Note that you should replace 'x' character by latest build number from this page For example, now it is build 1.1.0.13, so you should replace 'x' by '13': wget...

Good day. We fixed the issue with AAC to Opus transcoding while capturing RTSP with stereo audion and playing it as WebRTC in build 5.2.764. Please update and check.

Good day. We do not recommend to use WSL because WSL 1.0 provides no virtualization. We also do not recommend to use WSL 2 which is based on Linux kernel on Hyper-V becuase it ir really slow. So please try to use classic VM (VirtualBox or VMWare) with Ubuntu. By the way, you do not need Linux to...

You can wrap websocket session establishing to try-catch, for example try { session = Flashphoner.createSession(sessionOptions); } catch (WebsocketNotConnectedException e) { mPublishStatus.setText("Websocket connection failed") }

Each broadcast creates a load on the channel corresponding to the bitrate. Bandwidth calculations are very simple 1 broadcast with a bit rate of 1Mbps takes 1 Mbps of a communication channel. For bad channels, you need to reduce the bit rate and stream resolution. If necessary, you can use...

No, we did not touch server setup. We tested the case on our AWS instances: TURN server (internal and external on the same instance) works with relay policy disabled and does not work with relay policy enabled, both for manually deployed instance and AWS Marketplace one. Without load balancer...

Скорее всего не поддерживает кодек H.264 Известные проблемы п. 6

Good day. This means application cannot establish websocket connection to WCS. Please check if WCS server you're trying to connect to is up and running, and Websocket ports 8080 and 8443 is not blocked.

Good day. Please read this article about minimal code sampe for screen sharing. To record screen share stream, you should add record: true option to createStream() function call: session.createStream({ name: "mystream", display: document.getElementById("screensharing"), constraints...

Результат микширования, в нашем случае, это один суммированный сигнал, который нельзя поделить, т.е. одна видео дорожка в потоке и одна аудиодорожка. Если мы правильно понимаем задачу, вы хотите добавить к одному видеостриму несколько звуковых дорожек и переключаться между ними на стороне...

Good day. Probably yes, but not sooner than iOS SDK will migrate to Swift (ticket WCS-2853). We raised the ticket WCS-2882, but this take a long time. Now, we're working on ticket WCS-2794 to update libwebrtc which should fix the issues with RTCEAGLVideoView.

Good day. We successfully published stream using credentials you provided Please note that we do not use relay option: - Using this code, stream is published Flashphoner.createSession({ urlServer: 'wss://webrtc-dev.********.***:8443', mediaOptions: { "iceServers": [{...