Thank you, I really appreciate ! I will read that carefully.
FYI, we just did the test of sending the INVITE with transport=tcp in
contact header and it just works fine: kamailio sends the BYE over TCP.
Regards,
Pascal
On Wed, Apr 14, 2010 at 3:36 PM, Juha Heinanen wrote:
> Klaus Darilion writ
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/tut
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.
i tried to add alias usage example in sip router wiki, but was not able
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_
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.mailingli...@pernau.at> wrote:
> The contact after fix_nated_contact() should also contain ;transport=tcp.
> Thus, Kamailio should relay the BYE with TCP.
>
> Can yo
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
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
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
