Hi Stephan,
It is best to let SIPp generate and assign the Call ID so that SIPp
can track the individual calls. You can do this by using the [call_id]
key word. I'm also not sure if you are using the concept of the
remote_ip/port correctly - it is not really used to populate the "to"
address field header.
As the second question, I have not tested the following scenario, but
it may work with a little tweaking:
===============START OF FILE==============
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="UAC Basic Invite, then Cancel">
<label id="1" />
<!-- In client mode (sipp placing calls), the Call-ID MUST be
generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500" start_rtd="true">
<![CDATA[
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: John <sip:[EMAIL PROTECTED];tag=[call_number]-INV-UAC
To: Joey <sip:[EMAIL PROTECTED]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: John <sip:[EMAIL PROTECTED]:[local_port]>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=SIPp-UAC
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 response="100" rtd="true" optional="true" />
<recv response="180" rtd="true" optional="true" />
<recv response="400" rtd="true" rrs="true" next="3" optional="true" />
<recv response="401" rtd="true" rrs="true" next="3" optional="true" />
<recv response="403" rtd="true" rrs="true" next="3" optional="true" />
<recv response="404" rtd="true" rrs="true" next="3" optional="true" />
<recv response="406" rtd="true" rrs="true" next="3" optional="true" />
<recv response="408" rtd="true" rrs="true" next="3" optional="true" />
<recv response="480" rtd="true" rrs="true" next="3" optional="true" />
<recv response="486" rtd="true" rrs="true" next="3" optional="true" />
<recv response="487" rtd="true" rrs="true" next="3" optional="true" />
<recv response="500" rtd="true" rrs="true" next="3" optional="true" />
<recv response="503" rtd="true" rrs="true" next="3" optional="true" />
<recv response="504" rtd="true" rrs="true" next="3" optional="true" />
<recv response="200" rtd="true" rrs="true" next="2" />
<label id="2" />
<send>
<![CDATA[
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: John <sip:[EMAIL PROTECTED];tag=[call_number]-INV-UAC
To: Joey <sip:[EMAIL PROTECTED]>
Call-ID: [call_id]
CSeq: 5000 CANCEL
Contact: John <sip:[EMAIL PROTECTED]:[local_port]>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 0
]]>
</send>
<recv response="400" rtd="true" next="3" optional="true" />
<recv response="401" rtd="true" next="3" optional="true" />
<recv response="403" rtd="true" next="3" optional="true" />
<recv response="404" rtd="true" next="3" optional="true" />
<recv response="408" rtd="true" next="3" optional="true" />
<recv response="200" rtd="true" next="4" crlf="true" />
<label id="3" />
<nop>
<action>
<exec int_cmd="stop_call" />
</action>
</nop>
<label id="4" />
<!-- 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>
===============END OF FILE==============
Simon
On 9/7/07, Stephan Sutardi <[EMAIL PROTECTED]> wrote:
> Hello,
>
> i have two questions.
>
> 1.
> When i only edit the call-id in a fix number, sipp did not send a ACK?
>
> The Invite-Message is
> INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
> From: <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
> To: <sip:[EMAIL PROTECTED]:[remote_port]>
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 INVITE
> Contact: sip:[EMAIL PROTECTED]:[local_port]
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: [len]
>
> 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
> a=rtpmap:0 PCMU/8000
>
>
> The Call flow will be
> Message sent: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
> Message Received: SIP/2.0 100 Trying
> Message Received: SIP/2.0 180 Ringing
> Message sent: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
> Message Received: SIP/2.0 180 Ringing
> Message Received: SIP/2.0 180 Ringing
> Message Received: SIP/2.0 200 OK
> Message sent: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
> Message Received: SIP/2.0 200 OK
> Message sent: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
> Message Received: SIP/2.0 200 OK
> Message sent: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
> Message Received: SIP/2.0 200 OK
> Message sent: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
> Message Received: SIP/2.0 200 OK
>
> The Phone is waiting for an ACK-Message, but sipp doesn't send the
> message. And sipp only send six Invite messages.
> can you help me?
>
> The second question is:
> Is is possible to send only an Invite and a Cancel message?
> Can you send me the xml file?
>
> thx a lot
>
> Stephan
>
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