Hi Elrond,
Just some general pointers:
Add a [last_Record-Route:] to the 180 Ringing signal
Add a pause after the 180 Ringing for like 2 seconds!
Also ensure you don't have an XML well-formness error (SIPp is weak at
detecting them).
You scenario for example:
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="UAS Basic responder">
<recv request="INVITE" rrs="true" crlf="true" />
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]-INV-UAS
[last_Call-ID:]
[last_CSeq:]
Contact: [field0] <sip:[EMAIL PROTECTED]:[local_port]>
[last_Record-Route:]
Content-Length: 0
]]>
</send>
<pause milliseconds="2000" />
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]-INV-UAS
[last_Call-ID:]
[last_CSeq:]
[last_Record-Route:]
Contact: [field0] <sip:[EMAIL PROTECTED]:[local_port]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=SIPp-UAS
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 request="ACK" optional="true" crlf="true" />
<!-- 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. -->
<pause milliseconds="4000" />
<send retrans="500">
<![CDATA[
BYE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 4444 BYE
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true" />
...
...
Simon
On 10/20/07, Elrond Rules <[EMAIL PROTECTED]> wrote:
>
> Am trying to write a UAS scenario with the following as message seq:
> Sipp receives INV, sends 180, 200 OK and receives ACK
> After a pause, SIPP needs to send a BYE and receive a 200 OK (Sending a BYE
> is the deviation from the normal UAS scenario).
> I tried this with the following xml file and was not successful...
>
> <?xml version="1.0" encoding="ISO-8859-1" ?>
> <!DOCTYPE scenario SYSTEM "sipp.dtd">
> <scenario name="Basic Sipstone ">
>
>
> <recv request="INVITE" crlf="true">
> </recv>
>
> <send>
> <![CDATA[
>
> SIP/2.0 180 Ringing
> [last_Via:]
> [last_From:]
> [last_To:];tag=[call_number]
> [last_Call-ID:]
> [last_CSeq:]
> Contact:
> <sip:[local_ip]:[local_port];transport=[transport]>
> Content-Length: 0
>
> ]]>
> </send>
>
> <label id="1"/>
>
> <send retrans="500">
> <![CDATA[
>
> SIP/2.0 200 OK
> [last_Via:]
> [last_From:]
> [last_To:];tag=[call_number]
> [last_Call-ID:]
> [last_CSeq:]
> Contact:
> <sip:[local_ip]:[local_port];transport=[transport]>
> Content-Type: application/sdp
> Content-Length: 0
>
>
>
> 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 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=ftmp:101 1-11,16
> ]]>
> </send>
>
> <recv request="ACK"
> rtd="true"
> crlf="true">
> </recv>
>
>
> <!-- 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 sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
> Via: SIP/2.0/[transport] [local_ip]:[local_port]
> From: sipp
> <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
> To: sut
> <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
> Call-ID: [call_id]
> CSeq: 2 BYE
> Contact: sip:[EMAIL PROTECTED]:[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>
>
> What happens is when the sipp receives an invite it doesnt respond and when
> my invite times out from the remote end.
>
> And sipp throws an error saying "unexpected message while expecting a 200
> response"
>
> Do let me know what am I missing in this.
>
> thanks
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