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This is definitely the publisher channell quality issue. Please use TCP transport when publishing WebRTC stream from webcam session.createStream({ name: streamName, display: localVideo, constraints: { ... }, transport: "TCP" ... }).publish(); If this does not...

This message can be displayed by browser in two cases: 1. You are using self-signed certificate. 2. You are using Certificate Authority which is not trusted by the browser. So please do the following: 1. Check if you have succesfully imported the certificate using WCS dashboard. Is dashboard...

Also, these settings are recommended for most RTMP ingest points (Youtube, FB, Twitch etc): rtmp_transponder_full_url=true rtmp_transponder_stream_name_prefix= rtmp_flash_ver_subscriber=LNX 76.219.189.0 rtmp_transponder_send_metadata=true

Hello, This setting is required for broadcasting to YouTube: rtmp_flash_ver_subscriber=LNX 76.219.189.0 Please make sure that it is added to config flashphoner.properties.

Good day. We use WebRTC native library with some patches added, for example, to enable system audio capturing in screen capture extension. So SDK establishes signaling Websocket connection, passes stream publishing parameters to the library, and it does all the job: establishes WebRTC connection...

I'm sorry. Option 2 was. 2. Set permissions for folder and internal files cd /usr/local/FlashphonerWebCallServer/bin ./webcallserver set-permissions Anyway. Glad it works for you.

Hello Few troubleshooting options: 1. Run server in standalone mode to see output directly cd /usr/local/FlashphonerWebCallServer/bin ./webcallserver start standalone 2. Set permissions for folder and internal files cd /usr/local/FlashphonerWebCallServer/bin ./webcallserver set-permissions...

Измерения, проведенные с помощью Speedtest, к сожалению, не показатель. Эти измерения отображают пропускную способность канала до какого-то определенного сервера или серверов, размещенных в каком-то датацентре, но не до Вашего сервера WCS. Кроме того, здесь не отображаются потери пакетов...

Yes. Mentioned stream_mixer1 can be re-published as described here using REST API /rest-api/push/startup

Hello You can pull a conference if the conference is based on Mixer or MCU. https://docs.flashphoner.com/display/WCS52EN/Real-time+stream+mixer+with+MCU+functions Then you have exactly one mixed stream, that can be pulled as a simple RTMP stream. Example: 1. Create mixer /mixer/startup >...

No. Stream will be published using the transport defined only. You can implement fallback in client code: 1. Try to publish with transport: "UDP" 2. If STREAM_STATUS.FAILED with Failed by ICE timeout is received, try to publish againg with transport: "TCP"

Please try to use TURN server to bypass NAT issues at client side. Another option is to enable TCP transport for WebRTC (some providers block DTLS UDP traffic to prevent Tor access, but DTLS TCP is still working): session.createStream({ name: streamName, display: localVideo, ...

Низкий битрейт аудио и включенное эхоподавление будут приводить к низкому качеству звука: может булькать и прерываться. Рекомендуем аудио битрейт хотя бы 30000 и отключить эхоподавление audio: { bitrate: 30000, ... echoCancellation: false, googEchoCancellation: false, ... }...

Please see this post.

This means media ports (31001-32000/udp by default) on the server are unreachable. To fix this, check the following: 1. Are the ports opened for outgoing connections on client side? 2. If the server is not behind NAT, are the ports opened for incoming connections in firewall settings? 3. If the...

Good day. The page you mention is obsoleted. There is no any separate load testing tool for WCS now. Please read the following articles about actual load testing methods: What kind of server do you need to run a thousand WebRTC streams? Comparing load testing methods How many mixers can a...

No, this only means the problems with video encoding. This may be visual artifacts, freezes, lags, or video track may not be played at all. Please use the statistics page to check the server state and stream metrics to check a stream state.

Please check the following: 1. Are the instances to place behind the load balancer already running (step 1 of the manual)? Is the port 8081 accessible on every instance? 2. Is the load balancer security group created and configured (step 2.4 of the manual)? 3. Are the listener target groups...

Good day. Please clarify where is the first console.log[/ICODE} call you expect to reach in your code? If you expect to receive STREAM_STATUS.PUBLISHING event, but receiving STREAM_STATUS.FAILED, use [ICODE]stream.getInfo() method to get a reason to fail the publishing. Please see the source...

Good day. When you publish a WebRTC stream from Android SDK, VP8 codec is probably used (H264 is not supported on system level). In this case, video transcoding VP8->H264 is working on the server. But from browser with WebSDK H264 can be published because most browsers supports this codec out of...