On Mon, Jan 19, 2009 at 5:31 AM, Thomas Elsgaard
<[email protected]>wrote:
> Hi guys
>
> Does anyone have an XML scenario for an UAC registering, and then making
> one simple call, and then sending bye?
Hello,
try the attached file. I've just merged two scenarios that I have here.
It was not fully tested but it reached the point the INVITE is
authenticated.
>
> I only want to run the scenario once, how can i specify that in SIPp? i
> have only found the option to set the interval, like once per second
use "-m 1".
regards,
mayama
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Registration and call">
<!-- You must supply a file containing data for injection -->
<!-- field0 : remote username -->
<!-- field1 : username -->
<!-- field3 : domain -->
<!-- field3 : authentication -->
<!-- Sample
SEQUENTIAL
1234;gandalf;middle-earth.com;[authentication username=gandalf password=qerafecdsdr]
-->
<!-- arguments to sipp must include: -->
<!-- -i : local_ip -->
<!-- -p : local_port -->
<!-- -sf : scenario file -->
<!-- -inf : injection data file -->
<!-- -d : delay -->
<!-- Ex.: sipp -i 192.168.2.122 -p 6060 -sf test.xml -inf data.txt -d 20000 192.168.2.123 -->
<send retrans="500">
<![CDATA[
REGISTER sip:[field2] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
From: "[field1]" <sip:[fiel...@[field2]>;tag=[pid]SIPpTag00[call_number]
To: "[field1]" <sip:[fiel...@[field2]>
Call-ID: [call_id]
Contact: <sip:[fiel...@[local_ip]:[local_port]>
CSeq: 1 REGISTER
Max-Forwards: 70
Expires: 3600
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE
User-Agent: ATA
Content-Length: 0
]]>
</send>
<recv response="407" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[field2] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
From: "[field1]" <sip:[fiel...@[field2]>;tag=[pid]SIPpTag00[call_number]
To: "[field1]" <sip:[fiel...@[field2]>
Call-ID: [call_id]
Contact: <sip:[fiel...@[local_ip]:[local_port]>
CSeq: 1 REGISTER
[field3]
Expires: 3600
User-Agent: ATA
Content-Length: 0
]]>
</send>
<recv response="200">
</recv>
<send retrans="500">
<![CDATA[
INVITE sip:[fiel...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
From: "[field1]" <sip:[fiel...@[field2]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[fiel...@[remote_ip]:[remote_port]>
Call-ID: [call_id]
Contact: <sip:[fiel...@[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="407" auth="true">
</recv>
<send>
<![CDATA[
ACK sip:[fiel...@[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:[fiel...@[local_ip]:[local_port]>
CSeq: 801 ACK
Max-Forwards: 70
User-Agent: ATA
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[fiel...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
From: "[field1]" <sip:[fiel...@[field2]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[fiel...@[remote_ip]:[remote_port]>
Call-ID: [call_id]
Contact: <sip:[fiel...@[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:[fiel...@[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:[fiel...@[local_ip]:[local_port]>
CSeq: 803 BYE
[field3]
Max-Forwards: 70
User-Agent: ATA
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause/>
<!-- 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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