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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060
9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK sip:[EMAIL PROTECTED]:5060
10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE
sip:[EMAIL PROTECTED]:5060
11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE sip:[EMAIL PROTECTED]: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 <[EMAIL PROTECTED]> 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
> [email protected]
> 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:[EMAIL PROTECTED]:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0.
From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=22779SIPpTag001.
To: sut <sip:[EMAIL PROTECTED]:5060>.
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE.
Contact: sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060>;tag=22779SIPpTag001.
To: sut <sip:[EMAIL PROTECTED]:5060>.
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]:5060 out_uri=sip:[EMAIL PROTECTED]:5060
via_cnt==1".
.
#
U 1.2.3.4:5060 -> 4.3.2.1:5060
INVITE sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060>;tag=22779SIPpTag001.
To: sut <sip:[EMAIL PROTECTED]:5060>.
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE.
Contact: sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060>;tag=22779SIPpTag001.
To: sut <sip:[EMAIL PROTECTED]:5060>;tag=22700SIPpTag014.
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]:5060>;tag=22779SIPpTag001.
To: sut <sip:[EMAIL PROTECTED]:5060>;tag=22700SIPpTag014.
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]:5060>;tag=22779SIPpTag001.
To: sut <sip:[EMAIL PROTECTED]:5060>;tag=22700SIPpTag014.
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]:5060>;tag=22779SIPpTag001.
To: sut <sip:[EMAIL PROTECTED]:5060>;tag=22700SIPpTag014.
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-5.
From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=22779SIPpTag001.
To: sut <sip:[EMAIL PROTECTED]:5060>;tag=22700SIPpTag014.
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK.
Contact: sip:[EMAIL PROTECTED]:5060.
Max-Forwards: 70.
Subject: Performance Test.
Content-Length: 0.
.
#
U 1.2.3.4:5060 -> 4.3.2.1:5060
ACK sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060>;tag=22779SIPpTag001.
To: sut <sip:[EMAIL PROTECTED]:5060>;tag=22700SIPpTag014.
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK.
Contact: sip:[EMAIL PROTECTED]:5060.
Max-Forwards: 69.
Subject: Performance Test.
Content-Length: 0.
.
#
U 192.168.0.101:5060 -> 1.2.3.4:5060
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-7.
From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=22779SIPpTag001.
To: sut <sip:[EMAIL PROTECTED]:5060>;tag=22700SIPpTag014.
Call-ID: [EMAIL PROTECTED]
CSeq: 2 BYE.
Contact: sip:[EMAIL PROTECTED]:5060.
Max-Forwards: 70.
Subject: Performance Test.
Content-Length: 0.
.
#
U 1.2.3.4:5060 -> 4.3.2.1:5060
BYE sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060>;tag=22779SIPpTag001.
To: sut <sip:[EMAIL PROTECTED]:5060>;tag=22700SIPpTag014.
Call-ID: [EMAIL PROTECTED]
CSeq: 2 BYE.
Contact: sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060>;tag=22779SIPpTag001.
To: sut <sip:[EMAIL PROTECTED]:5060>;tag=22700SIPpTag014.
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]:5060>;tag=22779SIPpTag001.
To: sut <sip:[EMAIL PROTECTED]:5060>;tag=22700SIPpTag014.
Call-ID: [EMAIL PROTECTED]
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
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="1000">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: sut <sip:[EMAIL PROTECTED]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true" rrs="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
[routes]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause milliseconds="1000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="1000">
<![CDATA[
BYE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
[routes]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="500, 1000, 1500, 2000"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="500"/>
</scenario>
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uas' scenario. -->
<!-- -->
<scenario name="Basic UAS responder">
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv request="INVITE" crlf="true">
</recv>
<!-- The '[last_*]' keyword is replaced automatically by the -->
<!-- specified header if it was present in the last message received -->
<!-- (except if it was a retransmission). If the header was not -->
<!-- present or if no message has been received, the '[last_*]' -->
<!-- keyword is discarded, and all bytes until the end of the line -->
<!-- are also discarded. -->
<!-- -->
<!-- If the specified header was present several times in the -->
<!-- message, all occurences are concatenated (CRLF seperated) -->
<!-- to be used in place of the '[last_*]' keyword. -->
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
optional="true"
rtd="true"
crlf="true">
</recv>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<pause milliseconds="1000"/>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="200, 500"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="500, 1000, 2000, 3000"/>
</scenario>
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