[OpenSIPS-Users] Two OpenSIPS proxies issue

duane.larson at gmail.com duane.larson at gmail.com
Mon Jul 9 06:09:56 CEST 2012


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:








> 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.

















> 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.


> >


> >


> >


> >


> >


> > 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:


> >


> >


> > > Duane,


> >


> >


> > >


> >


> >


> > > 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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> > >


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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,


> >


> >


> > > 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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