15 May
2017
15 May
'17
4:29 p.m.
Hello Colin,
While I completely agree with you, this is what the client want, as he
already has this working with asterisk (which was patched to support this),
and he doesn't want to change the architecture... what can I say?
The tcp leg is on a private network, by the way.
David
On Mon, May 15, 2017 at 3:26 PM Colin Morelli <colin.morelli(a)gmail.com>
wrote:
Hi David,
This isn't an answer to your question - I'm not sure I have one for you
right now.
However, I've seen a few messages from you now about trying to get this
sips/sip stuff working, and I'm curious to know why you're jumping through
these hoops to make this work? The entire purpose of sips is that it should
be encrypted end to end. Is there a point in pretending to the client as
though they're using sips when they're actually using an insecure
connection in the middle? Is there a particular client you're trying to
cooperate with that needs this? You can still get TLS working over the
public side of the connection (using ;transport=tls in the SIP URI), while
allowing for insecure transports in your private network. This seems to be
ultimately what you want.
I'm sure you can get this to work, and I'm sure there's an answer - it
just seems like you're going to end up with something fairly fragile. It
seems like something you'd want to avoid unless you absolutely can't.
Best,
Colin
On Mon, May 15, 2017 at 9:20 AM, David Villasmil <
david.villasmil.work(a)gmail.com> wrote:
Hello guys,
I'm listening on port 443 for tls and 5111 for tcp.
Client registers on 443 and makes a call over tls which kamailio forwards
over tcp.
Because freeswitch doesn't support "sips" yet, when replies come back
from freeswitch (which contact set as "sip:" i manually change that tp
"sips:" for the client to not die with "SIPS Required".
Call is established and the client responds with ACK. At this point i
need to change back the contact "sips:" to "sip:".
This is the route that changes it:
route[FIXCONTACT] {
# This is freeswitch sending a message, so we change "sip:" to
"sips:"
xlog("L_ERR","[FIXCONTACT]: User-Agent is: $ua\n");
if( $ua =~ "^FreeSWITCH" ) {
if( subst('/^(Contact:.*)sip:/\1sips:/') ) {
xlog("L_ERR","[FIXCONTACT]: Method $rm Status $rs Changed
contact coming from freeSWITCH from sip to sips [$ct]!\n");
}
# Else it is the client, when change it back from "sips:" to
"sip:"
} else {
if( subst("/^(Contact:.*)sips:/\1sip:/") ) {
xlog("L_ERR","[FIXCONTACT]: Method $rm Status $rs Changed
contact coming from CLIENT from sips to sip [$ct]!\n");
}
}
return;
}
When kamailio is preparing to send out the ACK via TCP I'm seeing:
[forward.c:268]: get_send_socket2(): protocol/port mismatch
And I really don't know why it's doing this...
Help is appreciated!
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337 <+34%20669%2044%2083%2037>
ᐧ
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