Migration process is described below
Questions and Answers
Android Chrome browser, the call does not
work. Why?
How to migrate a WCS licene to another server?
I’m planning organization of a web broadcasting with a low latency. What is the latency value? What is a number of subscribers?
Web Call Server supports WebRTC technology. This means each client will establish WebRTC session with the server.
If a Flash-SIP call is established, can SIP user initiate a conference call and invite another SIP user?
Flash phone is able to participate in a conference which is established via a confbridge(PBX).
Is there way to broadcast a SIP stream to Wowza clients and record the stream?
The similar case is already implemented in master branch of client: https://github.com/flashphoner/flashphoner_client/tree/master
How does autologin work? Is it enough secure?
Autologin path is: WebRTC Client — WCS — Authentication server
When I place WebRTC call from
Android Chrome browser, the call does not
work. Why?
Android Chrome browser, the call does not
work. Why?
It seems that Android Chrome security subsystem requires valid CA signed certificate for DTLS.
RTP packets are not sent every 20ms exactly with RTMP SIP Gateway. Can we reduce the jitter?
As you know Flash sends the audio data via TCP stream.
