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Yes. Thank you for reply.

Hi admin, I have a call script that is called internal when a sip peer (8001) calls to extensions (999999) it goes to IVR, here will play audio. Then goes to queue (2000), the queue has music on hold (moh). Next connects to sip peer (8000). I tested SIP audio call and the video call does not...

I intend to follow my diagram because when setting connect both client and SIP server by IP local, I test make SIP video call does not have freeze frame. I tried to connect to wss of SIP server Asterisk to make SIP video call, it work betters. The opposite setup connects both client and SIP...

Thanks for your reply. I hope WCS has this function soon.

Before I tried setting options ip and ip_local by IP local and using http.address= IP public. I can register to SIP server and make calls but in the browser client can't send or receive rpt packet and after 6-7s this call auto

I tried this way, but WCS server using IP public connect to IP local of SIP server. It can't register to server althought I turn off firewalld.

I want to settup follow this diagram

If set options ip and ip_local same ip public or ip local it works when accessing the web demo by this. but if set all by ip local and http.address=ip public, access by ip public doesn't work.

My WCS server has an ip public that doesn't use NAT and I turned off the firewalld when testing.

Yes, it works for the public for demo interface but I intend for the call to have a path connection between SIP server and WCS server using IP local, a connection between WCS server and browser client using IP public when I make a call RTP traffic can't be sent to the browser client.

hi admin, I can setting WCS connect with SIP server by IP local and with browser run web demo by IP public? when I setting by options ip and ip_local in flashphoner.properites can registered to SIP server but call haven't RTP both audio and video

Thank you for your reply.

Hi admin. Can WCS automatically adjust image quality to network speed? If yes, how can i set it up?

I decreased Video FPS (at Phone Video) and switch TCP transport, but that problem still. Client use WIFI. What parameters can I change resolution/bitrate at flashphoner.properties?

Hi administrator, I'm using the feature SIP Phone Video, call from Browser to Browser situation occurs freeze-frame in the remote video. I debug in edge://webrtc-internals/. I see frame_Decoded_in_ms unstable. How can I improve this?

I see hook /call outgoing haven't parameter custom.

Thank you.

Can I open multiple WSS port on a server install WCS?

It's work. Thank you again.

Hi administrator, I'm using WCS version 5.2.1498. I change the default port at file flashphoner.properties with config: wss.port=443 (default 8443) and then restart webcallserver it OK, but I used the command netstat to check port 443 it's not listening. I'm certain port 443 hasn't been...