How does the Videxio browser-based solution work?
Our VMRs are integrated with unique access via browser-based video (WebRTC). This enables anyone with a Virtual Meeting Room (VMR) subscription from Videxio to invite internal and external participants to join a high quality video meeting by using their browser. There is no need to install anything on your computer or mobile device. The only requirement is to have an updated web browser (Chrome recommended), and a good local internet connection. Of course having an HD camera will always enhance the user experience!
This will enable seamless collaboration between the "web world" and traditional H323/SIP video conferencing solutions from vendors such as Cisco, Tandberg, Lifesize and Polycom, as well as other solutions such as Microsoft Lync.
Please note that WebRTC is still in development, and that the quality and functionality is continuously evolving and improving. To upgrade to the latest functionality, ensure that you always have an up-to-date browser.
Which web browsers are currently supporting WebRTC?
Please see this article.
Which devices are supported?
WebRTC is supported on:
PC/MAC running Windows, Mac OS or Linux.
Android devices running either Chrome version 29 or later
WebRTC is currently not supported on iOS devices.
Can the end user change the bandwidth settings?
Yes, the user does have the possibility to change the bandwidth settings manually by pressing the Settings button to the right of the Join Meeting button.
Quality of the connection:
- High – 1280kbps, up to 720p (HD)
- Medium – 768kbps, up to 480p (SD)
- Low – 384kbps, up to 288p (mobile)
The default value is set to Medium (768kbps connection).
What about presentation sharing, is it supported via WebRTC?
Participants connecting via WebRTC will be able to receive and send presentation using all the major browsers (Chrome, IE, Firefox, Opera, Safari).
How will the WebRTC client be able to traverse the firewall (tunnelling)?
By default browser checks if there are any UDP ports above 10000 open by the firewall. If not, both signalling and media traffic will use a single TCP port 443.
However, if the firewall has the below UDP ports open, then the browser will use:
- TCP port 443 for call signalling.
- UDP ports 1024-65535 for media traffic (audio, video and content)
Note: It is only required to open up ports from inside to outside.
Will the call from the WebRTC client be encrypted?
Yes, the signalling is encrypted using HTTPS and the media (if UDP ports are in use) via AES 128 bit encryption (SRTP).