On Wed, Apr 14, 2010 at 12:21 PM, Klaus Darilion < klaus.mailingli...@pernau.at> wrote:
> The contact after fix_nated_contact() should also contain ;transport=tcp. > Thus, Kamailio should relay the BYE with TCP. > > Oh I see, do you mean that the INVITE sent by A should includes ;transport=tcp in its Contact header ? Can you show an ngrep dump (ngrep -W byline -t -q -P "" port 5060) of the > problematic scenario? > > I'm not sure if I can share these traces :-/ I will see what I could do. Regards, Pascal > regards > klaus > > PS: A more standard-conform way of rewriting the SDP is to use the > add_contact_alias() and handle_ruri_alias() functions: > > http://sip-router.org/docbook/sip-router/branch/master/modules_k/nathelper/nathelper.html#id2601711 > > regards > klaus > > Am 14.04.2010 12:06, schrieb Pascal Maugeri: > >> Hi >> >> I need some "guidelines" to troubleshoot the following issue: >> >> a) A is behind NAT >> >> b) B is not behind NAT >> >> c) A calls B, SIP INVITE is sent over TCP >> >> d) A's firewall does NAT and changes the source port to let's say p1 >> >> e) B releases the call and sends BYE over UDP >> >> f) Kamailio sends the BYE to A, over UDP, to the NATed source port p1 ?! >> >> 2 comments: >> - This scenario works perfectly when A is the one who disconnects the >> call. and of course when no NAT is involved everything works ok >> - if works when I comment the line "fix_nated_contact()" in the >> route[NAT] block: >> >> route[NAT]{ >> #!ifdef WITH_NAT >> force_rport(); >> #if (nat_uac_test("19")) { >> if (nat_uac_test("3")) { >> if (method=="REGISTER") { >> setbflag(10); >> fix_nated_register(); >> } else { >> #fix_nated_contact(); >> [...] >> >> Any hint is very welcome. >> >> Cheers >> Pascal >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >
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