Hi Ben,
Thanks for answering this one. I agree totally with your analysis, but I can
also assure you that the "different" port number is present in the R-URI of the
ACK.
This is what OpenSIPS receives, over TLS, from the remote UAC:
ACK sip:+35312345...@195.xx.yy.zz:5050;transport=tcp SIP/2.0
I'm using port 5050 (tcp) on FreeSwitch and ports 5060 (tcp and udp) and 5061
(tls) on OpenSIPS.
I'm having to use a work-around at the moment that looks like this (slightly
edited):
if (has_totag() && is_method("INVITE|ACK|PRACK|BYE|UPDATE|REFER|NOTIFY")) {
$var(orig_ruri) = $ru;
$var(orig_rp) = $rp;
if ( loose_route() ) {
# Loose-routed request - attempt to match with an existing dialog
match_dialog();
if ($var(orig_rp) == "5050" && $ru =~ "r2=on") {
# Work-around fiddle for sending to an endpoint on the same IP
as myself
$ru = $var(orig_ruri);
$du = "sip:" + $rd + ":5050";
force_send_socket(tcp:195.XX.YY.ZZ:5060);
xlog("L_WARN", " LR(fixed) $rm RuProto=$rP R-URI=$ru
Dest=$du Call-ID=$ci\n");
} else
xlog("L_WARN", " LR $rm RuProto=$rP R-URI=$ru
Call-ID=$ci\n");
John Quick
Smartvox Limited
-----Original Message-----
From: Ben Newlin <ben.new...@genesys.com>
Sent: 19 May 2020 14:37
To: john.qu...@smartvox.co.uk; OpenSIPS users mailling list
<users@lists.opensips.org>; 'James Hogbin' <ja...@ip-sentinel.com>
Subject: Re: [OpenSIPS-Users] Running OpenSIPS on same box as
Asterisk/FreeSwitch
John,
This is occurring because the local OpenSIPS is seeing its own local URI in the
R-URI and is (correctly) assuming the message is meant for it. At that point
there is some ambiguity in the situation as even though the Route headers
indicate loose routing, OpenSIPS cannot leave the R-URI as is, because it will
just loop back to itself. So it is instead assuming that strict-routing is
being used and following that model.
We also run with different servers sharing IPs and we ran into this too. You
say the two servers are using different ports, but I am guessing that the R-URI
doesn't actually contain the port. That is why there is confusion. The only fix
I know is to ensure that every server sharing an IP always includes its local
port in the Contact header. This will cause all sequential requests to also
contain the port in their R-URI. This differentiates the R-URI so that OpenSIPS
will no longer believe the message is directed to it and will follow normal
loose routing procedures.
I'm not really sure whether there could be any improvement in OpenSIPS that
could handle this, since the situation is inherently ambiguous.
Ben Newlin
On 5/19/20, 3:56 AM, "Users on behalf of John Quick"
<users-boun...@lists.opensips.org on behalf of john.qu...@smartvox.co.uk> wrote:
Hi James and others,
James, I have provided some explanation and guidance specific to you at the
end of this post. First, I need to retract some of the explanation I gave in my
original post because it was wrong. The basic problem is as described -
handling of loose routed sequential requests goes wrong when OpenSIPS is
running on the same host server as Asterisk or FreeSwitch, using the same IP
but different ports.
I think the correct explanation for what is happening is this:
Where the address in the R-URI is the same as its local address, OpenSIPS
copies the URI (and parameters) from a Route header to the R-URI which, in this
use-case, causes the request to loop back to itself. It should instead delete
the two Route headers and leave the original R-URI value unchanged.
James, what you describe is not identical to my issue, although you may
have my issue plus another.
It should be fairly easy for you to fix the problem of private LAN
addresses being used where you want external IP's to be used.
OpenSIPS handles this issue through the "advertised address". This allows
it to listen on one interface, but to populate SIP headers such as Via and
Record-Route with another.
Various mechanisms are available to add to your script including the
following:
1) Define it for each interface in the listen statement: e.g. listen =
tls:10.x.x.x:5091 as <External IP>:5091
2) Global settings at the top of your script: advertised_address and
advertised_port
3) Within the main body of your script, e.g. just before the call to
record_route(), insert a line like: set_advertised_address(<External IP>);
4) Instead of calling record_route(), call
record_route_preset(<address1:port>,<address2:port>);
The first two are described in the section called 'Global Parameters' in
the documentation
The third one is described in 'Core Functions' (In v2.4, they are items 45
and 46)
The fourth is documented in the RR module notes
John Quick
Smartvox Limited
> From: James Hogbin <ja...@ip-sentinel.com>
> Sent: 18 May 2020 17:24
> To: john quick <john.qu...@smartvox.co.uk>; OpenSIPS users mailling list
<users@lists.opensips.org>
> Subject: Re: [OpenSIPS-Users] Running OpenSIPS on same box as
Asterisk/FreeSwitch
>
> I thought I had just done some bad config. I have almost exactly the same
thing.
>
> I have Freeswitch And Opensips on Azure VM’s
>
> Freeswitch = 10.0.0.4 + External IP
> Opensips = 10.0.0.5 + External IP
>
> Both are configured to use the external addresses
>
> Freeswitch will start the conversation on its external Gateway IP address
with the Opensips external IP address
>
> However the moment I put the Record_Route in the message to Teams
Freeswitch starts using the local 10.x.x.x. address on opensips rather than the
external address for ACK and BYE which rogers the whole thing up.
>
> I’m sure there is probably some way to fix it in the opensips plan as the
dialog/transaction ID’s are still there but it’s beyond me.
>
> James
> James Hogbin
> Director
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