Hello, Thanks for your explaination Sebastian but ... nothing change, RR has not been added. But i have noticed a new error: 400 Bad request (as you can see in my attachment).
Why there is so many ACK/200 OK on my wireshark trace. On my scenario, there is only one ACK no ? Is it normal ? Many thanks for your response ! 2015-07-06 11:37 GMT+02:00 Sebastian Damm <d...@sipgate.de>: > Hello, > > you are probably just sending the BYE to the wrong contact. The 200 OK > didn't come from your Kamailio but from the server where the request was > dispatched to. So you have to use the URI from Contact header as your new > Request URI. (This is true already for the ACK after the 200 OK was > received.) > > You have to change the sipp xml scenario file to > - extract the contact header and routes from the 200 OK > - include those information in every packet sent after the 200 OK. > > Keywords: [next_url], [routes] and "rrs" attribute in the recv command. > Help: http://sipp.sourceforge.net/doc/reference.html#UAC > > So the BYE should look like this: > > BYE [next_url] SIP/2.0 > [routes] > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] > From: sipp <sip:s...@loicchabert.fr>;tag=[pid]SIPpTag00[call_number] > To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] > Call-ID: [call_id] > CSeq: 2 BYE > Contact: sip:sipp@[local_ip]:[local_port] > Max-Forwards: 70 > Subject: Performance Test > Content-Length: 0 > > Hope that helps. > > Best Regards, > Sebastian > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
sip_error_sipp.pcapng
Description: application/pcapng
<?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. -->
<!--
Content-Type: application/sdp
-->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@loicchabert.fr SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:s...@loicchabert.fr>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[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="407" auth="true">
</recv>
<send>
<![CDATA[
ACK sip:[service]@loicchabert.fr SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: sipp <sip:s...@loicchabert.fr>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[service]@loicchabert.fr SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:s...@loicchabert.fr>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
[field2]
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>
<recv response="183" 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">
</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:[service]@loicchabert.fr SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:s...@loicchabert.fr>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[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/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE [next_url] SIP/2.0
[routes]
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:s...@loicchabert.fr>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[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="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
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