[OpenSIPS-Users] Two OpenSIPS proxies issue

Vlad Paiu vladpaiu at opensips.org
Wed Jul 11 10:40:31 CEST 2012


Hello,

Ok, thanks for the notification.
Seems I didn't look close enough at the config :D Anyway, glad that you 
got it fixed.

Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com


On 07/11/2012 05:50 AM, Duane Larson wrote:
>
> 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 
> <mailto: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 Developer
>     http://www.opensips-solutions.com  
>
>
>     On 07/09/2012 07:09 AM, duane.larson at gmail.com
>     <mailto: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 <mailto: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
>>     <mailto: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:
>>     >
>>     >
>>     >
>>     >
>>     >
>>     >
>>     >
>>     >
>>     > 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 <mailto: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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>>     > >
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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 <mailto: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 <mailto: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 <mailto:Users at lists.opensips.org>
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>>
>>     _______________________________________________
>>     Users mailing list
>>     Users at lists.opensips.org  <mailto:Users at lists.opensips.org>
>>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>     _______________________________________________
>     Users mailing list
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
> -- 
> --
> *--*--*--*--*--*
> Duane
> *--*--*--*--*--*
> --
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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