Hi List 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? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________ -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
