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 , [email protected] 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:[email protected]: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 , [email protected] 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.
>
>
>
>
>
> 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).
>
>
>
>
>
> I will send a PCAP trace to Ali directly.
>
>
>
>
>
>
>
>
> On , Ali Pey [email protected]> 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.
>
>
> >
>
>
> >
>
>
> > 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.
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > 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.
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > Regards,
>
>
> > Ali Pey
>
>
> >
>
>
> > On Sat, Jul 7, 2012 at 3:35 PM, osiris123d [email protected]>
wrote:
>
>
> >
>
>
> >
>
>
> > This is driving me crazy. I was right the first time when I said that
one of
>
>
> >
>
>
> >
>
>
> > the ACKs was not showing up as a loose route. It is the third ACK
that is
>
>
> >
>
>
> >
>
>
> > coming from the OpenSIPS/Proxy. When it reaches the OpenSIPS/SBC
device the
>
>
> >
>
>
> >
>
>
> > ACK fails as a loose route.
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > It would make sense that this would not be a loose route because
there are
>
>
> >
>
>
> >
>
>
> > no Route headers so the loose_route() function would return FALSE.
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > The issue still remains that when the ACK reaches the OpenSIPS/SBC it
still
>
>
> >
>
>
> >
>
>
> > isn't routed to the Callee, instead it is looped and routed to the
same
>
>
> >
>
>
> >
>
>
> > interface it came from because that is whats in the RURI.
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > --
>
>
> >
>
>
> >
>
>
> > View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/Two-OpenSIPS-proxies-issue-tp7580685p7580743.html
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> > _______________________________________________
>
>
> >
>
>
> >
>
>
> > Users mailing list
>
>
> >
>
>
> >
>
>
> > [email protected]
>
>
> >
>
>
> >
>
>
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> >
>
>
> >
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