No, I don't have re-invite... I think.
I atatch my scenario.
Thanks for your help :)
On Tue, Nov 24, 2009 at 9:33 AM, Evgeny Miloslavsky <
[email protected]> wrote:
> Do you have Re-INVITe procedure within your scenario? If you do, it might
> couse described problem
>
>
>
> Regards,**
>
> *Evgeny Miloslavsky*
>
> *Systest Engineer*
>
> *Juniper Networks Solutions Israel LTD.*
>
> *Office: 972-9-9712355*
>
> *Office: 972-74-7170072***
> ------------------------------
>
> *From:* Vanesa Tejada [mailto:[email protected]]
> *Sent:* Tuesday, November 24, 2009 10:09 AM
> *To:* Evgeny Miloslavsky
> *Cc:* [email protected]
>
> *Subject:* [Sipp-users] Scenario blocked and don't finished.
>
>
>
> Uhm... yesterday I was checking the logs... I notice that It is sent 2
> INVITES and only one of them is processing, then the scenario is block
> waiting messages for the other INVITE.
> I removed the resend period, but I couldn't resolv it :S
>
> How can I send only 1 INVITE? I use "-m 1" in line command.
>
>
> Thanks
>
> I'm working on SuSe Linux and SiPp 3.1
>
>
> On Tue, Nov 24, 2009 at 7:38 AM, Evgeny Miloslavsky <
> [email protected]> wrote:
>
> I think it happens because of –nd option
>
>
>
> Regards,
>
> *Evgeny Miloslavsky*
>
> *Systest Engineer*
>
> *Juniper Networks Solutions Israel LTD.*
>
> *Office: 972-9-9712355*
>
> *Office: 972-74-7170072*
> ------------------------------
>
> *From:* Vanesa Tejada [mailto:[email protected]]
> *Sent:* Monday, November 23, 2009 1:01 PM
> *To:* SipP List
> *Subject:* [Sipp-users] Scenario blocked and don't finished.
>
>
>
> Hi,
>
>
> I wanted to know how a scenario can be blocked.
>
> I run a simplescenario, I send the ACK at the end of scenario then the the
> scenario label is closed </scenario>, but I don't know why, sipp continues
> running before ACK and don't finish never.
>
> Does anybody knows why that happend?
>
>
> Thanks in advanced
>
>
> --
> Vanessa Tejada
>
>
>
>
> --
> Vanessa Tejada
>
>
--
Vanessa Tejada
<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic Sipstone UAC">
<send>
<![CDATA[
INVITE [asimpu] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 70
Route: <sip:[email protected]:6060;lr>
From: "[field0]"<sip:[fiel...@[field1]>;tag=[call_number]
To: <[asimpu]>
Call-ID: [call_id]
CSeq: 1 INVITE
Supported: 100rel
Contact: sip:[fiel...@[local_ip]:[local_port]
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
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
]]>
</send>
<recv response="100" optional="true">
</recv>
<!--Salto a PRACK-->
<recv response="183" rrs="true" next="1">
<action>
<!--Hemos de quedarnos con el RSeq del 183 para formar el RAck en PRACk-->
<ereg regexp=".*" search_in="hdr" header="RSeq: " assign_to="1"/>
</action>
</recv>
<recv response="180" optional="true">
<!--Salto a recibir 200 del final!! next="2"-->
</recv>
<label id="1"/>
<!--pause milliseconds="1000"/-->
<!--PRACK-->
<send>
<![CDATA[
PRACK [next_url] SIP/2.0
[last_Via:]
Max-Forwards: 70
[routes]
From: "[field0]" <sip:[fiel...@[field1]>;tag=[call_number]
[last_To:]
[last_Call-ID:]
CSeq: 2 PRACK
RAck: [$1] 1 INVITE
Expires: 3600
Contact: sip:[fiel...@[local_ip]:[local_port]
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
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos mandatory remote sendrecv
]]>
</send>
<!-- 200 OK about PRACK -->
<recv response="200" rrs="true">
</recv>
<!--send>update</send-->
<send>
<![CDATA[
UPDATE [next_url] SIP/2.0
[last_Via:]
Max-Forwards: 70
[routes]
[last_From:]
[last_To:]
[last_Call-ID:]
CSeq: 3 UPDATE
Contact: <sip:[fiel...@[local_ip]:[local_port];transport=[transport]>
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
a=curr:qos local sendrecv
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos mandatory remote sendrecv
]]>
</send>
<!-- 200 OK about UPDATE-->
<recv response="200" rrs="true">
</recv>
<!-- <label id="2"/> -->
<!-- 200 OK about 183 -->
<recv response="200" rrs="true">
</recv>
<!--Segun el enunciado aqui podriamos empezar la session-->
<!--SEND ACK about INVITE-->
<send>
<![CDATA[
ACK [next_url] SIP/2.0
[last_Via:]
Max-Forwards: 70
[routes]
From: "[field0]" <sip:[fiel...@[field1]>;tag=[call_number]
[last_To:]
[last_Call-ID:]
CSeq: 1 ACK
Content-Length: 0
[last_Contact:]
]]>
</send>
<ResponseTimeRepartition value="1000, 5000, 20000"/>
<CallLengthRepartition value="1000, 5000, 20000"/>
</scenario>
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