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Unfortunately, VOD does not support pausing at all. You can only start VOD live stream from the beginning of the file or stop it. A workaround is to split VOD file to a small fragments and play them subsequentally, in this case you can stop playback and start it again from a closest fragment...

Good day. The sample file contains only webm header and no media data (has 0:00:00 duration). This can occur if there was no media data in the stream published, or no key frames. The recording can start only from key frame, so seems like ther were no keyframes in the stream during this time.

Good day. You can use REST API: VOD capture management with REST API /vod/startup to start capture from VOD /mixer/add to add VOD stream to mixer /vod/terminate to stop VOD stream

Добрый день. В дальнейшем, пожалуйста, направляйте отчеты или ссылки на них только с помощью формы. Сообщение пришлось удалить, т.к. отчет, доступный по прямой ссылке, может содержать данные лицензии. В логах видно, что перед перезапуском остановилась запись только одного потока, причем только...

Now, most of streaming services require to send metadata. But, if there is no metadata sent, some media servers (Wowza, WCS for example) can get picture parameters directly from media packets. However, seems like Youtube does not support this. So you have to disable RTMP buffering to workaround...

We made some tests almost an year ago. Up to 10 streams 320x240 may be played on the same page on Intel Core i5 8gen (8 HT cores) with 20 Gb RAM. But you have to play WebRTC: MSE can be played only 1 stream per session, so it is required to set up SSL certificates and open the page via HTTPS. If...

We tested republishing from your server. There are two issues in server configuration: 1. The following parameter transcoding_disabled=true prevents any transcoding. But Opus audio must be transcoded to AAC for republishing, so audio is not sent to RTMP server. Please enable transcoding...

Good day. We successfully tested WebRTC as RTMP republishing on your server. It's working, stream is publishing and republishing to our demo server: Please make sure you're opening the page using HTTPS (this is the only case Webrtc will work). Also please make sure you're using a correct stream...

Good day. The first stream we see in logs provided seems to be captured successfully But you're opening the page via HTTP. not HTTP, so WebRTC does not work Please use HTTPS to open a page in browser. Also, we can't reproduce the problem in Player example...

Please also clarify: do you play 15 streams simyltaneously: 1. from one page in browser? 2. from one workstation (PC/Mac) in 15 tabs of one browser? 3. from 15 workstation (one strem per one PC)? For both cases 1 and 2, seems local CPU resources are not enough to play, so local browser...

Sorry, it's a mistype. Should be sudo /usr/local/FlashphonerWebCallServer/bin/webcallserver check_update and sudo /usr/local/FlashphonerWebCallServer/bin/webcallserver update respectively. Please see the doc here.

Hello How exactly do you play 15 streams. We need a code sample, screenshots, full browser console logs in text form. 5 lines of js logs are definitely not enough.

Good day. Please reproduce the issue in Two Way Streaming example https://wcs:8444/client2/examples/demo/streaming/two_way_streaming/two_way_streaming.html. If this is not reproduced, please modify the example code minimally to reproduce the issue and send the code using this form.

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...