[OpenSIPS-Users] Two OpenSIPS proxies issue

Wed Jul 11 04:50:07 CEST 2012

Just to update...

My config on the OpenSIPS/SBC was all jacked up.  I basically have two
routes in my config, one for SIP messages coming from the LAN and one for
SIP messages coming from the WAN.  On the LAN side before I was doing my
"if has_totag" and "if loose_route" I had unintentionally put my "if method
is not REGISTER|MESSAGE then record_route()" before it.  Then on my WAN
side I didn't even have the "if method is not REGISTER|MESSAGE then
record_route()".  So that was really jacking up my routing.  Not having the
"if method is not REGISTER|MESSAGE then record_route()" in my WAN side
route was the reason why I saw the INVITE coming from the OpenSIPS/Proxy
and the Record_route headers not being in the correct order when my Callee
received the INVITE relayed by the OpenSIPS/SBC.

Also like Ali said my contacts were also an issue.  I saw Jeff's post about
fix_contact() so I got rid of that on my OpenSIPs/Proxy device.

Things look a lot better now.  I thought all that duct taping was hidding
something and it got out of control.

Thanks for working with me.  This really gave me a good refreshers course
in SIP routing.

On Mon, Jul 9, 2012 at 3:45 AM, Vlad Paiu <vladpaiu at opensips.org> wrote:

> **
> Hello,
>
> This is quite strange, can you please also post a full OpenSIPS debug for
> the call where that ACK got relayed out like
>
>         ACK
> sip:50.xx.xx.156;lr;ftag=d4xut7i3jx;nat=yes;vst=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA;did=0f9.1ddb82a6
> SIP/2.0.
>         Record-Route: <sip:99.xx.xx.161;r2=on;lr>.
>         Record-Route: <sip:192.168.88.1;r2=on;lr>.
>
>
> Regards,
>
> Vlad Paiu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
>
> On 07/09/2012 07:09 AM, duane.larson at gmail.com wrote:
>
> I just got my calls working by removing the Record-Route's and then
> reinserting then in an order that would according to my topology.
>
> I will need to go back and start from scratch to see if a lot of the other
> stuff I did was really needed or not and then update but here is were I
> edited the Record-Routes
>
> When the INVITE is coming from my OpenSIPS/Proxy to the Callee I did
>
> if ( is_method("INVITE") ) {
> remove_hf("Record-Route");
> insert_hf("Record-Route: $(hdr(Record-Route)[2])\r\n", "Via");
> insert_hf("Record-Route: $(hdr(Record-Route)[1])\r\n", "Via");
> insert_hf("Record-Route: $(hdr(Record-Route)[0])\r\n", "Via");
> }
>
> Then when the 180 and 200 are coming from the Callee to the Caller before
> the 180 and 200 go to the Caller I did the following
>
>
> if (t_check_status("180")){
> remove_hf("Record-Route");
> insert_hf("Record-Route: $(hdr(Record-Route)[2])\r\n", "Via");
> insert_hf("Record-Route: $(hdr(Record-Route)[1])\r\n", "Via");
> insert_hf("Record-Route: $(hdr(Record-Route)[0])\r\n", "Via");
>
> }
>
>
> if (t_check_status("200")){
> remove_hf("Record-Route");
> insert_hf("Record-Route: $(hdr(Record-Route)[2])\r\n", "Via");
> insert_hf("Record-Route: $(hdr(Record-Route)[1])\r\n", "Via");
> insert_hf("Record-Route: $(hdr(Record-Route)[0])\r\n", "Via");
>
> }
>
>
> So not sure if there is something wrong with the way OpenSIPS places the
> Record-Route ordering when OpenSIPS has multiple interfaces. I am not 100%
> sure if what I have done here is right or not but calls are working now.
>
> Any feedback?
>
>
> On , duane.larson at gmail.com wrote:
> > I think I have multiple issues going on but I might be getting closer to
> the issue.
> >
> >
> >
> >
> >
> > I am wondering if this might be part of the issue.
> >
> >
> >
> >
> >
> > If you look at the the following,
> http://www.tech-invite.com/Ti-sip-dialog.html#inv , for the first INVITE
> message that the Callee receives the first Proxy that the callee needs to
> take in its Record-Route is first in the list of Record-Routes on the
> INVITE message. As for the Caller the Record-Route set gets flipped
> (whatever Record-Route is on the top will be its last route hop). So if
> this is the case then why is the OpenSIPS/SBC device sending my Callee
> device an INVITE message with the far end proxy, OpenSIPS/Proxy, on the top
> of the Record-Route list? Here is the INVITE that my callee is getting
> >
> >
> >
> >
> >
> > INVITE sip:9013XX3XX6 at 192.168.88.14:3072;line=9zx0whnm SIP/2.0
> >
> >
> > Record-Route:
> 4aoni525hc;nat=yes;vst=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA;did=598.b8b26331>
> >
> >
> > Record-Route:
> >
> >
> > Record-Route:
> >
> >
> >
> >
> >
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> >
> >
> > The Record-Route with 50.XX.XX.156 should be at the bottom of the list I
> think because that is the OpenSIPS/Proxy that is on the Internet. Am I
> wrong on this? On the SIP trace I posted on pastebin this INVITE to the
> Callee starts on line 299.
> >
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> > On , duane.larson at gmail.com wrote:
> >
> >
> > > I'm really not sure if I am just duck taping the issue but I was able
> to make most of the call work. The only problem now is when the Callee
> hangs up the BYE is sent directly to the OpenSIPS/Proxy IP instead of going
> to the OpenSIPS/SBC. This will not work due to firewall issues.
> >
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> > > My ACKs are no longer not showing up as Non-Loose Route messages, but
> the BYEs are.
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> > > So if the Caller hangs up the Callee sees the BYE message (GOOD!), but
> if the Callee hangs up the Caller never sees the BYE message (Bad).
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> > > I will send a PCAP trace to Ali directly.
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> > > On , Ali Pey alipey at gmail.com> wrote:
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> > > > Duane,
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> > > > The Ack should not have any request-route headers. Only Route
> headers. If you see request-route headers, then you need to find how they
> got there and fix that first.
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> > > > I believe it is ok if the Ack doesn't go through loose route, in
> that case it should be sent to the request-uri destination ip and that IP
> should be your client IP.
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> > > > Let me know if this help. If not, can you attach here a wireshark
> trace and I will go through your signalling for you. Going thought a text
> trace can be quit time consuming. In wireshark it's a lot easier to jump
> from a message to another through the call flow. You can use tcpdump to
> capture to .cap file for wireshark.
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> > > > Regards,
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> > > > Ali Pey
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> > > > On Sat, Jul 7, 2012 at 3:35 PM, osiris123d duane.larson at gmail.com>
> wrote:
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> > > > This is driving me crazy.  I was right the first time when I said
> that one of
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> > > > the ACKs was not showing up as a loose route.  It is the third ACK
> that is
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> > > > coming from the OpenSIPS/Proxy.  When it reaches the OpenSIPS/SBC
> device the
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> > > > ACK fails as a loose route.
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> > > > It would make sense that this would not be a loose route because
> there are
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> > > > no Route headers so the loose_route() function would return FALSE.
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> > > > The issue still remains that when the ACK reaches the OpenSIPS/SBC
> it still
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> > > > isn't routed to the Callee, instead it is looped and routed to the
> same
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> > > > interface it came from because that is whats in the RURI.
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> > > > --
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> > > > View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Two-OpenSIPS-proxies-issue-tp7580685p7580743.html
> >
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> > > > Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
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> > > > _______________________________________________
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> > > > Users mailing list
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> > > > Users at lists.opensips.org
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> > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>

-- 
--
*--*--*--*--*--*
Duane
*--*--*--*--*--*
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