Search results

We do not provide ETA on forum according to EULA To prevent application hangs or crashes while repeately clicking join/leave consider to mute user actions until session is established as described here

If you're publishing RTMP stream using RTMP encoder (ffmpeg, OBS, Xsplit, Vix etc) please make sure that: 1. Keyframes are sending regularly (for example, ever 2 seconds) 2. GOP size is mutiple of FPS (for example, if you're publishing 30 fps, set GOP size 60 3. Stream bitrate fits to publisher...

We investigated the issue. Please add the following parameter to /usr/local/FlashphonerWebCallServer/conf/flashphoner.properties file rtmp_flash_ver_subscriber=LNX 76.219.189.0 and restart WCS. Republishing to Yuotube will work. The full republishing settings list...

With the settings recommended in this post, two ports will be used for SIP signaling alternately: 30000 and 30001. The last port in range must be even, and it is never used. So two ports is the minimum you can configure.

We answered you in this topic

You can use Room.publish() method to publish both camera and screen sharing stream. This example shows how to capture screen and publish it using Session.createStream() and Stream.publish(). You can take the screen capture source code, then integrate it to Conference example and use...

The minimum tested Ubuntu version is 18.04. Please also refer to this Linux distribution packages list.

Reproduced the issue (republish to YouTube fails) - internal ticket WCS-3031. Will notify when there is an update.

Reproduced the issue (app stops responding after several join-leave when there is another participant) - internal ticket WCS-3030. Will notify when there is any update.

You can send it using this form.

sudo systemctl restart webcallserver

Hello, Please see these Android SDK examples - Video Conference: doc, code - Screen Sharing: doc, code

Good day. We checked your instance. Seems like ip parameter was incorrectly set to local IP address. It is automatically set to external IP address on every WCS startup, so after server restart publishing and playback seems working:

You should use a properly configured certificates for WebRTC to work, especially on mobile devices. So you have to get SSL certificates from Let'sEncrypt for development purposes, or test strictly in desktop Chrome browsers confirming security exception. We checked publishing and playback, it...

Happy New Year! Unfortunately, there are no news yet, due to more priority tasks. We'll let you know here about any news.

Good day. Please check if you imported a valid SSL certificates for your domain as described here

Please check if you've restarted WCS after adding these strings. To test republishing to Facebook using WebRTC as RTMP example, do the following: 1. Create translation in FB Live producer 2. Open WebRTC as RTMP example page. Set Stream field to test and RTMP URL to Server URL and Stream Key...

Good day. Please read Video Chat and Screen Sharing example description and its code on GitHub. Please note that all the modern browsers now can capture screen without extension.

We checked your stream and server. Unfortunately, there is no any movement in the stream, just black screen with one white point both in VLC and in Media Devices example The stream does not fail during a couple of minutes, packets received count is growing, lost packets count is zero, stream...

Please use the following settings: port_from=30000 port_to=30002 In this case, port 30001 will be used for SIP signaling, and even ports (30000 and 30002) will not be used.