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Good day. A mixer created by REST API /mixer/startup may stop due to a various reasons: - if there are no incoming streams in mixer in 60 seconds (by default) - if mixing fails due to lack of CPU resources - etc So to find an exact reason you should collect a report as described here using...

Good day. Please read this article: What kind of server do you need to run a thousand WebRTC streams? Only a test as described in the article may give exact answer. Approximately, it seems like m5.2xlarge can handle up to 400 incoming WebRTC streams. But, if you're publishing VP8 (from Android...

No. You can only check for updates manually sudo /usr/local/FlashphonerWebCallServer/bin check_update and update if needed sudo /usr/local/FlashphonerWebCallServer/bin update But it is not recommended to update server in production unless it's not working as expected.

We recommend to test with one RTMP server (where the issue is reproducing) to get a cleaner logs to detect a reason. For example: - republishing to other server: works - republishing to Youtube: does not work. In this case, the disconnection reason will be in flashphoner.log. For example, if...

The CPU consumption issue is fixed in build 5.2.1252. Seems like you're using an earlier build. Please update WCS to the latest 5.2.1322, it should help.

We checked your server. Seems like you're trying to republish the same stream to 2 RTMP ingest points: Youtube and other streaming server. Please try to republish the stream to one RTMP server only. Also, you're explicitly setting republishing resolution to 1920x1080. In this case, stream will...

Hello Each forum thread has button "Report". You can click there to open the private email form for passing ssh. Or directly open the link &required=ssh Regarding RTMP. How do you test? Using REST API or using direct session.createStream().publish(); as in the WebRTC as RTMP example...

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, ... }...