2012/2/20 Henrik Aagaard Sørensen <henrikaagaardsoren...@gmail.com>
> I'm just trying to figure out how to use SIPp for the very first time,
> so I hope this question is OK.
>
> I've tried to read the documentation, but I cannot find any example on
> a very basic test.
>
> I would like to simulate the following:
>
> SIP-device: USERNAME:PASSWORD
> SIP-server: SIP.MY-DOMAIN.COM
> Number to call: 1234
>
> How can I have the sip-device calling the sip-server to number 1234
> for 20 simultaneous calls for about 10 minutes?
>
Hello,
sipp is very well documented. Everything is explained here:
http://sipp.sourceforge.net/doc/reference.html
But to get you started, try the attached files.
Call sipp like this:
sipp -i local_ip -p local_port -sf uac.xml -inf uac.txt -m 20 -d 600000
server_ip:server_port
regards,
Takeshi
SEQUENTIAL
1234;USERNAME;MY-DOMAIN.COM;[authentication username=USERNAME
password=PASSWORD];calling_number
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="uac with auth">
<!-- You must supply a file containing data for injection -->
<!-- field0 : remote username -->
<!-- field1 : local username -->
<!-- field2 : domain -->
<!-- field3 : authentication -->
<!-- field4 : CallingNumber -->
<!-- Sample
SEQUENTIAL
001234;1001;test.com;[authentication username=1001 password=1001];0312341234
-->
<!-- Call it like this: sipp -i 192.168.1.1 -p 5060 -sf uac.xml -inf uac.txt -d 600000 -m 20 192.168.1.2 -->
<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: "[field4]" <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: "[field4]" <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>
------------------------------------------------------------------------------
Try before you buy = See our experts in action!
The most comprehensive online learning library for Microsoft developers
is just $99.99! Visual Studio, SharePoint, SQL - plus HTML5, CSS3, MVC3,
Metro Style Apps, more. Free future releases when you subscribe now!
http://p.sf.net/sfu/learndevnow-dev2
_______________________________________________
Sipp-users mailing list
Sipp-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sipp-users
