On Tue, Mar 29, 2011 at 1:59 AM, michal javorka <kostr...@gmail.com> wrote:
> Hi,
> I am trying to test an opeser proxy server, i am using this tutorial
> http://www.troubleshootingwiki.org/OpenSER#Stress_Test_The_SIP_Signaling
> i found this in the text: "In the sample XML files of SIPp, record-routing
> is not supported. Please change the script accordingly." i dont know how to
> change them i was trying it long time, can anybody give me suggestion how to
> do this?
>
Try the attached scenario. It is the one I use.
HTH.
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="ATA origination">
<!-- You must supply a file containing data for injection -->
<!-- field0 : remote username -->
<!-- field1 : local username -->
<!-- field2 : domain -->
<!-- field3 : authentication -->
<!-- Sample
SEQUENTIAL
001234;1001;test.com;[authentication username=1001 password=1001]
-->
<!-- arguments to sipp must include: -->
<!-- -i : local_ip -->
<!-- -p : local_port -->
<!-- -sf : scenario file -->
<!-- -inf : injection data file -->
<!-- -d : delay (call duration) in milliseconds -->
<!-- Ex.: sipp -i 192.168.2.122 -p 6060 -sf uac.xml -inf data.txt -d 1000 192.168.2.123 -->
<send retrans="500">
<![CDATA[
INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
From: "[field1]" <sip:[field1]@[field2]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field0]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
Contact: <sip:[field1]@[local_ip]:[local_port]>
CSeq: 801 INVITE
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE
Supported: replaces
Content-Type: application/sdp
User-Agent: ATA
Content-Length: [len]
v=0
o=CMI-SIPUA 61838 0 IN IP[local_ip_type] [local_ip]
s=SIP CALL
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
]]>
</send>
<recv response="401" auth="true" next="auth_required" optional="true"/>
<recv response="407" auth="true" next="auth_required"/>
<label id="auth_required" />
<send>
<![CDATA[
ACK sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-2]
[last_From:]
[last_To:]
[last_Call-ID:]
Contact: <sip:[field1]@[local_ip]:[local_port]>
CSeq: 801 ACK
Max-Forwards: 70
User-Agent: ATA
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
From: "[field1]" <sip:[field1]@[field2]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field0]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
Contact: <sip:[field1]@[local_ip]:[local_port]>
CSeq: 802 INVITE
[field3]
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE
Supported: replaces
Content-Type: application/sdp
User-Agent: ATA
Content-Length: [len]
v=0
o=CMI-SIPUA 61838 0 IN IP[local_ip_type] [local_ip]
s=SIP CALL
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio 60000 RTP/AVP 0 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
]]>
</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" rrs="true">
</recv>
<send>
<![CDATA[
ACK [next_url] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch-5]
[last_From:]
[last_To:]
[last_Call-ID:]
[routes]
Contact: <sip:[field1]@[local_ip]:[local_port]>
CSeq: 802 ACK
[field3]
User-Agent: ATA
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
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch-1]
[last_From:]
[last_To:]
[last_Call-ID:]
[routes]
Contact: <sip:[field1]@[local_ip]:[local_port]>
CSeq: 803 BYE
[field3]
Max-Forwards: 70
User-Agent: ATA
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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