Solved Understanding Webclient traffic

[rdaeseleer]

Hi,

If I connect to 3CX using only the webclient, do I still need some priority in place for voice traffic?
Do I still have SIP/RTP traffic when all my clients are connected through the webclient? When I do a capture, I do not see any SIP/RTP traffic towards the 3CX. I only see some STUN and UDP traffic coming FROM the 3CX server to the client, is this normal behaviour? I would expect to see also some traffic generated from the client to the 3CX server since I also generate speach and that needs to be delivered to the other side..how is that traffic send to the 3CX server?

Thanks in advance
Randy

 

[Evolute IT]

All that uses the WebRTC protocol and not SIP, thus why you don't see the traffic. It is technically a Websocket over HTTPS with RTP support. It all works through regular Web protocols.

 

[rdaeseleer]

Hi Frederick,

Thank you for your quick reply. When I started a call I see some STUN packets and a lot of UDP traffic from 3CX server to client. Is this STUN for letting the client know which port to use? The ports used for the UDP stream are all in the range for 3CX Media Server. Is this the audiostream of the call that is send to the client? If somehow this is delayed, would that give poor audio quality and 'robotic' voice?

Thanks in advance
Randy

 

[ChrisC_3CX]

@rdaeseleer

Just to add to what @Frederick Marcoux said, STUN is used for NAT traversal purposes and it will allow the client to discover its public IP address and the port number. As you correctly said, the ports used on the 3CX PBX end for the UDP stream are in the same range as that used by the 3CX Media Server(9000-10999). Specifically, WebRTC with 3CX uses the following range for the RTP traffic: 10500-10999. This is indeed the audiostream and anything affecting it will most likely affect the audio quality as well.

 

[rdaeseleer]

Hi Chris, Frederick,

Thank you for your answers. This was the information I was looking for.
Regards,
Randy