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Hi, I have an WCS at AWS and I can't login to the admin account with the user/pass of admin/<ec2-instance-id> This instance is created from an AMI that was created from an AWS marketplace WCS. [1] How can I login to the admin account? [2] Is there a way to replace the current SSL certificate...

Hi I'm using WCS 5.2.945 with internal TURN server. I publish rtmp video and play it on web-rtc over TURN. The issue: From time to time the video on the web-rtc player stuck, sometimes for 2 seconds and sometimes for more than 5 seconds (at the same time the video played fine on the rtmp...

I'm trying to run WCS 5.2.912 on AWS EC2 in order to try reproduce an issue we had. I tried to do it in 2 ways: [1] Use an instance of older/newer version available in AWS marketplace and upgrade it to 912 [2] Use a clean ubuntu instance and install 5.2.912 on it. In both cases, I end up with...

Hi, I'm trying to pass audio from my server to the client via WCS. Publish rtsp opus audio Play webrtc over turn I sometime succeed to play the audio and sometimes fail. WCS version 5.2.945 I've attached flashphoner log from my last case. In this case I'm publishing rtsp stream, failed on the...

Hi! I'm using WCS 5.2.945 I publish an rtsp video stream to the server by using its dns name rtsp://<flashphoner-dns-name>:554/live/my-stream I want to play it with web-rtc and rtmp players and I only able to play when I provide to the player the stream name with the direct server ip...

Hi, I'm running WSC 5.2 on AWS from marketplace. WSC opens ports and listens on the private ip. I can't access to WSC ports (8444 for example) via the public ip. What should I do? Thanks!

Hi, I have a strange phenomenon while playing webrtc over turn. I'm publishing rtmp video h264 + acc, 6 fps, to flashphoner WCS and play it in several methods: 1. WebRTC - plays well. 2. ffplay (to play the rtmp stream) - plays well. 3. WebRTC over TURN (internal server) - Has issues: When the...

Hi, I'm using WSC 5.2 from marketplace on AWS and I'm trying to use the internal TURN server and I'll appreciate some help here. It seems that the TURN server failed to listen on port 3478 on the external interface. I followed this document...

Hi! I new to this server and I'm using Web Call Server 5 trial version. I'm trying to send my stream as rtsp to the server using ffmpeg. My command line is: ffmpeg -loglevel verbose -re -i music.m4a -c:a aac -ac 2 -ar 48000 -b:a 96K -f rtsp -rtsp_transport tcp rtsp://server-ip:554/live/test.sdp...