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Please update JDK to 12 or 14 and set up ZGC as described here (the example describes 24 Gb memory heap, you should adopt this to 32 Gb). Also please check CPU load while testing. Read this article for details what other metrics are important and how to monitor them. Please also read this...

Good day. In the latest builds, WCS starts from non-root user for better security. But ubuntu base docker image, which is used to reduce WCS image size, contains no sudo. So we raised the ticket to allow launching WCS from root in docker environment WCS-2997, and let you know results in this...

Good day. The REST hooks /connect and /playStream combination is the only way to count and limit the number of every stream subscribers. You can also apply ACL key to the stream if you're using CDN. In this case, you should send ACL key from backend to subscribers you want to allow to see the...

Good day. In 5.2 builds, stream recording is significally changed under the hoods to provide more stability and performance. Now, recording file name is known only when recording is finished, because recording file name template can contain timestamp fields. So stream.getRecordName() returns a...

Yes. Someone send the RSTP URL via "Report" button from your thread. Sorry for incovinuence.

Hello Supported format for RTSP video: H.264 (MPEG4 part 10) audio: AAC or G.711 https://docs.flashphoner.com/display/WCS52EN/From+an+IP+camera+via+RTSP MJPEG/SXGA is not supported. Try to configure h264. It should work.

Good day. Regarding stream rtsp://*****:*****@**.***.***.21:554/h264/ch1/main/av_stream. This RTSP source send video in MotionJPEG codec: v=0 o=- 2208990726 2208990736 IN IP4 127.0.0.1 s=Video Server Session c=IN IP4 0.0.0.0 t=0 0 a=control:* a=range:npt=0- m=video 0 RTP/AVP 26...

Good day. WSPlayer uses PCMU audio codec, it is only suitable for speech. Please consider to use WebRTC (Opus audio codec) or HLS (AAC audio) to play streams in Safari browser on Mac and iOS, in this case you'll get a better sound quality.

Настроили на вашем Windows сервере тестовый стенд. На стенде удалось получить около 2000 потоков без деградации контрольного стрима. Рекомендуем отключить TCP транспорта на CDN серверах на время тестирования. ice_tcp_transport=false TCP помогает при потерях, но дает задержку, поэтому внутри...

The admin password must be unique, this is AWS EC2 requirement, so we changing admin password to instance Id on first server start after installation.

After clean istallation, you must activate your license number, so please enter the license key when web interface asks it. Then, you could login as admin. Also, we've sent our public IP to you through forum conversations. If you're having trouble with installation, please provide SSH access to...

Добрый день! По всей видимости у вас плохой канал не между серверами в датацентре, а между сервером и клиентом. Рекомендую проверить с помощью Iperf ширину канала между серверами и рабочей станцией, на которой запускалось тестирование. Дополнительно, нужно на тестирующем сервере...

Good day. We reproduced the issue and raised the ticket WCS-2996 to investigate it. Will let you know results in this topic

Добрый день. Вы можете запустить трансляцию VOD-live из файла POST /rest-api/vod/startup HTTP/1.1 HOST: 192.168.1.101:8081 content-type: application/json { "uri":"vod-live://sample.mp4", "localStreamName": "sample_file" } и затем добавить этот поток в микшер POST /rest-api/mixer/add...

Good day. Yes, but you have to configure WCS and write some client side code: 1. Configure the server for publishing quality check inbound_video_rate_stat_send_interval=1 then restart server 2. While publiushing a stream, handle CONNECTION_QUALITY.UPDATE event and republish the stream with...

According to screenshot in this post, your server uses DTLS 1.0 while establishing WebRTC connection. This protocol is fully removed from Chrome 87. So we recommend to update WCS to latest build from this page to support DTLS 1.2 by default. If you're using REST hooks, please also check if your...

Good day. The only way is to play stream within the same websocket session like Two Way Streaming or Media Devices examples. In Google Cloud, you can enable stickiness by client IP, in this case balancer should route all the connections from the same IP to the same instance, but this is bad...

It won't work production ready. You can see dirty workaround in the post above for MSE or WSPlayer https://forum.flashphoner.com/threads/about-using-webrtc-from-non-ssl-web-screen.13292/post-25626 Another way you can try to add two streams into the mixer (one stream). REST API /mixer/startup...

Looks like server-side issue. Please provide ssh access to server using this private form. We will check. Or send report using report.sh script and private form. Try service webcallserver restart Or Check and configure ip and ip_local settings in...

Palakis/obs-websocket This OBS WebRTC implementation is not supported. Only supported implementation is https://docs.flashphoner.com/display/OBS10EN You can also use OBS RTMP