Connect Application is based on WebRTC technology. This means that the connections and bandwidth traffic are mainly handled by client side only. 


The traffic on the web application server is not high, as the connections are peer-to-peer. Although as soon as you start reaching outside your network – into a corporate firewall, for example – you're going to need a little more, well, firepower. Firewall configurations won't let WebRTC in without using the STUN (Session Traversal Utilities for NAT) or TURN (Traversal Using Relays around NAT) protocol. This is why you'll need a different server, called STUN / TURN Server. And this STUN / TURN Server needs to be able to handle your stream's bandwidth.


Firewall configurations won't let WebRTC in without using the STUN (Session Traversal Utilities for NAT) or TURN (Traversal Using Relays around NAT) protocol. This is why you'll need a server.


For one to one meeting, the number of concurrent sessions can be unlimited. For multi user video meeting, the number of members in meeting is limited by client-side resources such as bandwidth, memory and processor limitations to process large amount of video and audio streams. Broadcast type meeting such as Audio Broadcast (Podcast) or Video Broadcast (Webinar, Live Classes etc.), there is one or two broadcaster and many viewers that are subscribed to the streaming video and audio.


Types of Meetings

  •     Video Conference - where every member of the meeting can see, talk and listen to other members of the meeting. It is a Mesh Topology, every member is transmitting video and audio to & from every other member.
  •     Audio Conference - where every member of the meeting can talk and listen to other members of the meeting. It is same as Video Conference but only audio is transmitted, so less bandwidth required.
  •     Podcast - where Initiator of the meeting can broadcast their audio to other members of the meeting and other members can only listen to Initiator's audio, but not each others.
  •     Webinar - where Initiator of the meeting can broadcast their audio & video to other members of the meeting and other members can only see listen to Initiator's video & audio, but not each others.
  •     Live Class - where Teacher (Initiator) can see and listen to all students, students can see and listen to teacher but not other students.


So to summarise, You can cram anywhere from one to a million users into a WebRTC call. It all depends of the WebRTC Architecture you are using, type of meeting, quality of the stream, internet speed and bandwidth etc.


These are the links of some of articles that you can read to know and learn more about the WebRTC and related terms:


WebRTC Architectures Explained in 5 Minutes or Less

https://www.callstats.io/blog/webrtc-architectures-explained-in-5-minutes-or-less


WebRTC Multiparty Video Alternatives, and Why SFU is the Winning Model

https://bloggeek.me/webrtc-multiparty-video-alternatives/


How Many Users Can Fit in a WebRTC Call?

https://bloggeek.me/how-many-users-webrtc-call/


What is the required server configuration?

https://kodemintserviceshelp.freshdesk.com/support/solutions/articles/81000319096-what-is-the-required-server-configuration-