On Wed, Apr 14, 2010 at 12:21 PM, Klaus Darilion < klaus.mailinglists@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/n...

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

IMO the client is broken as it announces in Contact header that it want's to be contacted via UDP, although it is sending the message via TCP.

As suggest use add_contact_alias() instead of fix_nated_contact() to add the protocol always to the Contact URI. Of course then you have to use handle_ruri_alias() on all in-dialog requests.

regards klaus

Am 14.04.2010 13:53, schrieb Pascal Maugeri:

Klaus

I attached the ngrep you asked me to this email.

Regards, Pascal

On Wed, Apr 14, 2010 at 12:21 PM, Klaus Darilion <klaus.mailinglists@pernau.at mailto:klaus.mailinglists@pernau.at> wrote:

The contact after fix_nated_contact() should also contain
;transport=tcp. Thus, Kamailio should relay the BYE with TCP.

Can you show an ngrep dump (ngrep -W byline -t -q -P "" port 5060)
of the problematic scenario?

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



    _______________________________________________
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    mailing list
    sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
    http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Klaus Darilion writes:

As suggest use add_contact_alias() instead of fix_nated_contact() to add the protocol always to the Contact URI. Of course then you have to use handle_ruri_alias() on all in-dialog requests.

ok, i managed to get the wiki page done:

http://sip-router.org/wiki/tutorials/alias-example

-- juha