On Sat, Dec 6, 2008 at 3:17 PM, cool goose <[EMAIL PROTECTED]> wrote:
> Thank You very much. It works!! Is there a better tutorial on SIPp other
> than the documentation on their site? Can you suggest any resources that can
> help me in writing my own custom scenarios using SIPp ?
I think the online documentation is very good. That's from where I learned
it.
You will not need much beyond that.
Also, you can search the forum archives:
http://sourceforge.net/mailarchive/forum.php?forum_name=sipp-users
Other than that, use
sipp -sd uac
sipp -sd uas
etc.
to get the embedded sample uac, uas scenarios.
Also, I'm sending some scenarios that I use here to test sip proxies.
But to write scenarios is really easy, just capture a call using
ngrep/sipgrep/wireshark. Then put the messages from one of the peers in the
xml file and start replacing the variant info with the adequate
placeholders.
<?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:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
From: "[field1]" <sip:[EMAIL PROTECTED]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[EMAIL PROTECTED]:[remote_port]>
Call-ID: [call_id]
Contact: <sip:[EMAIL PROTECTED]:[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:[EMAIL PROTECTED]:[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:[EMAIL PROTECTED]:[local_port]>
CSeq: 801 ACK
Max-Forwards: 70
User-Agent: ATA
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
From: "[field1]" <sip:[EMAIL PROTECTED]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[EMAIL PROTECTED]:[remote_port]>
Call-ID: [call_id]
Contact: <sip:[EMAIL PROTECTED]:[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:[EMAIL PROTECTED]:[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:[EMAIL PROTECTED]:[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>
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="BGS as destination">
<!-- arguments to sipp must include: -->
<!-- -i : local_ip -->
<!-- -p : local_port -->
<!-- -sf : scenario file -->
<!-- Ex.: sipp -i 192.168.2.121 -p 6060 -sf uas.xml -->
<!-- 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>
<send>
<![CDATA[
SIP/2.0 100 Trying
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_From:]
[last_Call-ID:]
[last_CSeq:]
Contact: Excel<sip:[EMAIL PROTECTED]:[local_port]>
[last_Via:]
User-Agent: Excel_CSP/83.10.226
Content-Length: 0
]]>
</send>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_From:]
[last_Call-ID:]
[last_CSeq:]
Contact: Excel<sip:[EMAIL PROTECTED]:[local_port]>
[last_Via:]
User-Agent: Excel_CSP/83.10.226
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_From:]
[last_Call-ID:]
[last_CSeq:]
Contact: Excel<sip:[EMAIL PROTECTED]:[local_port]>
[last_Record-Route:]
Supported: timer
Session-Expires: 21600; refresher=uas
[last_Via:]
User-Agent: Excel_CSP/83.10.226
Content-Type: application/sdp
Content-Length: [len]
v=0
o=sip 1220546354 1220546354 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 98
a=sendrecv
a=rtpmap:98 telephone-event/8000
]]>
</send>
<!-- The [last_Via:] above will cause all Via Headers received in the last message to be added at that point in the SIP message, but they will be combined in a single Header (separated by commas). So the message will not be exactly the same as generated by a real CSP -->
<recv request="INVITE" optional="true" />
<!-- the above is used to ignore duplicated INVITE sent by OpenSER -->
<recv request="ACK"
crlf="true">
</recv>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_To:]
[last_From:]
[last_Call-ID:]
[last_CSeq:]
[last_Via:]
User-Agent: Excel_CSP/83.10.226
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. -->
<timewait milliseconds="4000"/>
<!-- 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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