23 Feb
2007
23 Feb
'07
10:38 a.m.
I Just re-read the docs on loose_route(). So please disregard this
question. ( only processed if Route: header is present. Which isn't
present because Record-route: header isn't being sent to caller )
So, I'm still trying to figure out why record-route: header is not
being sent to caller.
On 2/22/07, Andy Pyles <andy.pyles(a)gmail.com> wrote:
Hi Bogdan,
After running additional debugs, for some reason the call to
loose_route() is failing.
if (loose_route()) {
# mark routing logic in request
xlog("L_INFO", "loose_route() succeeded\n ");
route(1);
} else{
xlog("L_INFO", "loose_route()failed - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
};
Any ideas why this could be occuring?
On 2/22/07, Andy Pyles <andy.pyles(a)gmail.com> wrote:
HI Bogdan,
I'm already using an almsot identical version of uas.xml and uac.xml (
yes rrs=true) is being used. However in your version the uas.xml
doesn't have rrs="true" after initial invite which I think is needed.
See as you can see below, setting rrs="true" for uac will only work if
it receives a Record-Route header in the 200OK which it's not.
In this case, ALL messages from openser to sipp uac do not contain the
Record-route header. So I don't think it's a sipp problem, but an
openser configuration problem. I've tried using other devices for a
uac, such as x-lite but the same problem.
Andy
On 2/22/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote:
Hi Andy,
so it's about sipp :D - I remember I had some hard times to make it work
with record Route.
take a look at the attached files, they might help you.
regards,
bogdan
Andy Pyles wrote:
HI Bogdan,
thanks for your reply.
yes you are correct. The Bye doesn't have the Route header.
It appears the the 200 OK sent to the caller doesn't contain a
Record-route header.
Messages between openser and callee contain record-route information,
but messages between caller and openser do not.
Is there a way to enable that?
Here's more detail:
192.168.0.101 = Caller (sipp)
1.2.3.4 = openser
4.3.2.1 = callee ( sipp)
1.) 192.168.0.101 -> 1.2.3.4 SIP/SDP Request: INVITE
sip:service@1.2.3.4:5060, with session description
2.) 1.2.3.4 -> 192.168.0.101 SIP Status: 100 Giving a try
3.) 1.2.3.4 -> 4.3.2.1 SIP/SDP Request: INVITE
sip:service@4.3.2.1:5060, with session description
4.) 4.3.2.1 -> 1.2.3.4 SIP Status: 180 Ringing
5.) 4.3.2.1 -> 1.2.3.4 SIP/SDP Status: 200 OK, with session
description
6.) 1.2.3.4 -> 192.168.0.101 SIP Status: 180 Ringing
7.) 1.2.3.4 -> 192.168.0.101 SIP/SDP Status: 200 OK, with session
description
8.) 192.168.0.101 -> 1.2.3.4 SIP Request: ACK
sip:service@1.2.3.4:5060
9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK sip:service@4.3.2.1:5060
10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE
sip:service@1.2.3.4:5060
11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE sip:service@4.3.2.1:5060
12.) 4.3.2.1 -> 1.2.3.4 SIP Status: 200 OK
13.) 1.2.3.4 -> 192.168.0.101 SIP Status: 200 OK
---
Packets 6,7 and following contain no Record-route information.
The other weird thing is that openser is passing on the Route: header
it recevied from callee to the caller.
Please see attached for complete ngrep output.
On 2/21/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote:
> Hi Andy,
>
> could you check on the net if the BYE contain the Route hdr added to
> INVITE as Record-Route? I have some doubts on this as I see:
> 0(966) find_first_route: No Route headers found
> 0(966) loose_route: There is no Route HF
>
> and if the BYE is not identified, the dialog is not closed.
>
> regards,
> bogdan
>
> Andy Pyles wrote:
> > Hello,
> >
> > I have a question on how to configure the dialog module ( 1.2.x from
> > cvs yesterday ).
> >
> > With my config, ( attached) I can make calls and have verified that
> > the acc module is working correctly.
> >
> > My question is, when I enable the dialog module, I can see that it is
> > incrementing call count correctly, but when a bye is received, the
> > dialog:active_dialogs statistic is never decremented.
> >
> > In the debug level 9 logs, ( also attached) I see this error after the
> > 200OK is sent to the bye:
> >
> > 1(969) DBUG:dialog:unref_dlg: unref dlg 0xa7ce5a98 with 1
> (delete=0)-> 1
> >
> > Is this a case of one of the timers being set too short? by the way
> > using a variable call length from well under a second ( using sipp )
> > to 20 second call doesnt' seem to make a difference .
> >
> >
> > Thanks,
> > Andy
> >
> ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Users mailing list
> > Users(a)openser.org
> > http://openser.org/cgi-bin/mailman/listinfo/users
> >
>
>
------------------------------------------------------------------------
interface: lo (127.0.0.0/255.0.0.0)
filter: (ip or ip6) and ( port 5060 )
#
U 192.168.0.101:5060 -> 1.2.3.4:5060
INVITE sip:service@1.2.3.4:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0.
From: sipp <sip:sipp@192.168.0.101:5060>;tag=22779SIPpTag001.
To: sut <sip:service@1.2.3.4:5060>.
Call-ID: 1-22779(a)192.168.0.101.
CSeq: 1 INVITE.
Contact: sip:sipp@192.168.0.101:5060.
Max-Forwards: 70.
Subject: Performance Test.
Content-Type: application/sdp.
Content-Length: 137.
.
v=0.
o=user1 53655765 2353687637 IN IP4 192.168.0.101.
s=-.
c=IN IP4 192.168.0.101.
t=0 0.
m=audio 6000 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.
#
U 1.2.3.4:5060 -> 192.168.0.101:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0.
From: sipp <sip:sipp@192.168.0.101:5060>;tag=22779SIPpTag001.
To: sut <sip:service@1.2.3.4:5060>.
Call-ID: 1-22779(a)192.168.0.101.
CSeq: 1 INVITE.
Server: OpenSer (1.2.0-pre6-notls (i386/linux)).
Content-Length: 0.
Warning: 392 1.2.3.4:5060 "Noisy feedback tells: pid=22746 req_src_ip=192.168.0.101
req_src_port=5060 in_uri=sip:service@1.2.3.4:5060 out_uri=sip:service@4.3.2.1:5060
via_cnt==1".
.
#
U 1.2.3.4:5060 -> 4.3.2.1:5060
INVITE sip:service@4.3.2.1:5060 SIP/2.0.
Record-Route: <sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74>.
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKa6e8.b1ea5ac3.0.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0.
From: sipp <sip:sipp@192.168.0.101:5060>;tag=22779SIPpTag001.
To: sut <sip:service@1.2.3.4:5060>.
Call-ID: 1-22779(a)192.168.0.101.
CSeq: 1 INVITE.
Contact: sip:sipp@192.168.0.101:5060.
Max-Forwards: 69.
Subject: Performance Test.
Content-Type: application/sdp.
Content-Length: 137.
.
v=0.
o=user1 53655765 2353687637 IN IP4 192.168.0.101.
s=-.
c=IN IP4 192.168.0.101.
t=0 0.
m=audio 6000 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.
#
U 4.3.2.1:5060 -> 1.2.3.4:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKa6e8.b1ea5ac3.0.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0.
From: sipp <sip:sipp@192.168.0.101:5060>;tag=22779SIPpTag001.
To: sut <sip:service@1.2.3.4:5060>;tag=22700SIPpTag014.
Call-ID: 1-22779(a)192.168.0.101.
CSeq: 1 INVITE.
Route: <sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74>.
Contact: <sip:4.3.2.1:5060;transport=UDP>.
Content-Length: 0.
.
#
U 4.3.2.1:5060 -> 1.2.3.4:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKa6e8.b1ea5ac3.0.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0.
From: sipp <sip:sipp@192.168.0.101:5060>;tag=22779SIPpTag001.
To: sut <sip:service@1.2.3.4:5060>;tag=22700SIPpTag014.
Call-ID: 1-22779(a)192.168.0.101.
CSeq: 1 INVITE.
Route: <sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74>.
Contact: <sip:4.3.2.1:5060;transport=UDP>.
Content-Type: application/sdp.
Content-Length: 125.
.
v=0.
o=user1 53655765 2353687637 IN IP4 4.3.2.1.
s=-.
c=IN IP4 4.3.2.1.
t=0 0.
m=audio 6000 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.
#
U 1.2.3.4:5060 -> 192.168.0.101:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0.
From: sipp <sip:sipp@192.168.0.101:5060>;tag=22779SIPpTag001.
To: sut <sip:service@1.2.3.4:5060>;tag=22700SIPpTag014.
Call-ID: 1-22779(a)192.168.0.101.
CSeq: 1 INVITE.
Route: <sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74>.
Contact: <sip:4.3.2.1:5060;transport=UDP>.
Content-Length: 0.
.
#
U 1.2.3.4:5060 -> 192.168.0.101:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0.
From: sipp <sip:sipp@192.168.0.101:5060>;tag=22779SIPpTag001.
To: sut <sip:service@1.2.3.4:5060>;tag=22700SIPpTag014.
Call-ID: 1-22779(a)192.168.0.101.
CSeq: 1 INVITE.
Route: <sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74>.
Contact: <sip:4.3.2.1:5060;transport=UDP>.
Content-Type: application/sdp.
Content-Length: 125.
.
v=0.
o=user1 53655765 2353687637 IN IP4 4.3.2.1.
s=-.
c=IN IP4 4.3.2.1.
t=0 0.
m=audio 6000 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.
#
U 192.168.0.101:5060 -> 1.2.3.4:5060
ACK sip:service@1.2.3.4:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-5.
From: sipp <sip:sipp@192.168.0.101:5060>;tag=22779SIPpTag001.
To: sut <sip:service@1.2.3.4:5060>;tag=22700SIPpTag014.
Call-ID: 1-22779(a)192.168.0.101.
CSeq: 1 ACK.
Contact: sip:sipp@192.168.0.101:5060.
Max-Forwards: 70.
Subject: Performance Test.
Content-Length: 0.
.
#
U 1.2.3.4:5060 -> 4.3.2.1:5060
ACK sip:service@4.3.2.1:5060 SIP/2.0.
Record-Route: <sip:1.2.3.4;lr=on;ftag=22779SIPpTag001>.
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKa6e8.b1ea5ac3.2.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-5.
From: sipp <sip:sipp@192.168.0.101:5060>;tag=22779SIPpTag001.
To: sut <sip:service@1.2.3.4:5060>;tag=22700SIPpTag014.
Call-ID: 1-22779(a)192.168.0.101.
CSeq: 1 ACK.
Contact: sip:sipp@192.168.0.101:5060.
Max-Forwards: 69.
Subject: Performance Test.
Content-Length: 0.
.
#
U 192.168.0.101:5060 -> 1.2.3.4:5060
BYE sip:service@1.2.3.4:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-7.
From: sipp <sip:sipp@192.168.0.101:5060>;tag=22779SIPpTag001.
To: sut <sip:service@1.2.3.4:5060>;tag=22700SIPpTag014.
Call-ID: 1-22779(a)192.168.0.101.
CSeq: 2 BYE.
Contact: sip:sipp@192.168.0.101:5060.
Max-Forwards: 70.
Subject: Performance Test.
Content-Length: 0.
.
#
U 1.2.3.4:5060 -> 4.3.2.1:5060
BYE sip:service@4.3.2.1:5060 SIP/2.0.
Record-Route: <sip:1.2.3.4;lr=on;ftag=22779SIPpTag001>.
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK76e8.ecad2904.0.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-7.
From: sipp <sip:sipp@192.168.0.101:5060>;tag=22779SIPpTag001.
To: sut <sip:service@1.2.3.4:5060>;tag=22700SIPpTag014.
Call-ID: 1-22779(a)192.168.0.101.
CSeq: 2 BYE.
Contact: sip:sipp@192.168.0.101:5060.
Max-Forwards: 69.
Subject: Performance Test.
Content-Length: 0.
.
#
U 4.3.2.1:5060 -> 1.2.3.4:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK76e8.ecad2904.0.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-7.
From: sipp <sip:sipp@192.168.0.101:5060>;tag=22779SIPpTag001.
To: sut <sip:service@1.2.3.4:5060>;tag=22700SIPpTag014.
Call-ID: 1-22779(a)192.168.0.101.
CSeq: 2 BYE.
Route: <sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74>.
Contact: <sip:4.3.2.1:5060;transport=UDP>.
Content-Length: 0.
.
#
U 1.2.3.4:5060 -> 192.168.0.101:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-7.
From: sipp <sip:sipp@192.168.0.101:5060>;tag=22779SIPpTag001.
To: sut <sip:service@1.2.3.4:5060>;tag=22700SIPpTag014.
Call-ID: 1-22779(a)192.168.0.101.
CSeq: 2 BYE.
Route: <sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74>.
Contact: <sip:4.3.2.1:5060;transport=UDP>.
Content-Length: 0.
.
exit
39 received, 0 dropped