Video conference issue with specific network


#1

Hello,

I have an issue where users from a certain network are having issues establishing a video conference. Here’s the sequence of events.

User A: (With problematic network; corporate guest wifi network)
User B: (Any network)

Use Case:

  1. User A creates, joins a room, and waits for user B to join.
  2. User A can see his/her own video while waiting.
  3. As soon as user B joins, his/her own video stream disappears and replaces it with a broken image (I’m assuming the avatar image)
  4. User A and B never enter into a video conference

NOTE(s):

  • On user A’s computer, if the network is switched, it will work just fine.
  • On user A’s console log, there are no network issues being reported before user B joins.
  • On user A’s console log, as soon as user B joins, there are xmpp ping timeouts.
  • This is not an issue with meet.jit.si; only on our server
  • Jitsi server is on Azure
    • Port TCP 443 is open
    • Port TCP 80 is open
    • Ports UDP 10000-20000 are open

#2

Not sure why would he see ping timeouts …
I suspect the reason to work on meet.jit.si is the fact we are running a turn server, which is used when needed in p2p connections and also to handle media over tcp if needed.


#3

Thanks, I will try it out.

As for debugging, are there more logging that we can turn on to make the web and/or services more verbose?

Thanks,
Mark


#4

You just need to check webrtc-internals to see what is going on with the media, maybe.
chrome://webrtc-internals.
See which are the candidates and the protocol user A is using when a successful call is established with meet.jit.si.
Also check the network tab in the developer console to see what are the ping errors, are they network errors or some error returned … something can reveal more to find it.


#5

If there are three participants does is still use webrtc? The reason I’m asking is that if user A (problematic network) is using meet.jit.si, it works just fine.

With my limited understanding of webrtc, I thought webrtc and turn only comes into play if it’s peer-to-peer (two people).

-Mark


#6

@markmadlangbayan webrtc is used everytime. P2P connection is established if possible and there are two participants, if it is not possible or more than two participants connection to the bridge is established.
Turn can be used for both the p2p connection and jvb connection. This said means that if there is a turn configured, p2p will always succeed.
If you disable on the client machine all udp and tcp traffic to the videobridge then the connection will go through the turn server when the jvb connection is needed.