[asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?
benoit.panizzon at imp.ch
Fri Dec 27 09:40:50 CST 2019
I wonder how SIP via TCP is supposed to work. Not realy Asterisk
related, but I hope you experts might be able to help out :-)
One of our customers has a SIP device registering via a complex NAT. To
benefit from TCP Connection Tracking, he choose TCP instead of UDP.
So he expected, that an incoming call would be sent back to him on the
already open TCP connection, making it easy to get through that NAT.
This is not the case. Our SBC is attempting to initiate a new SIP TCP
connection towards the NAT Firewall of the customer thus getting
dropped because this is not the outgoing established connection opened
during the registration.
So, how should SIP via TCP work? Should one TCP session be used for all
signaling of potentially multiple concurrent calls, as expected by our
customer. Or is it usual to make one TCP session per call as observed?
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