Well, you'd have another nickel from over here, then. <div><br></div><div>If I can get this working before I'm tasked with something else I'll write up something more on the wiki about "Freeswitch and SIPp", but I'm not sure I'll get that chance.</div>
<div><br></div><div><div>BB</div><div><br><div class="gmail_quote">On Tue, Aug 25, 2009 at 11:05 AM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">I wish I had a nickel for every guy struggling with sipp load testing vs real world traffic.<div><div></div><div class="h5">
<br><br><br><div class="gmail_quote">On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga <span dir="ltr"><<a href="mailto:tculjaga@gmail.com" target="_blank">tculjaga@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex">Hello Takeshi,<br><br>Thanks for your hint... it worked out... so to be precise:<br><br>VIA header of both INVITE and ACK messages MUST be identical (IP:PORT + branch)... and you are right... it might not be according to SIP specification. Anyhow, i get FS understand my ACK message.<br>
<br><br>Finally, here is what i used and I'm getting some poor results .. but this is another topic :)<br><br><br>Thanks for your help.<br>Tihomir.<br><br><br>sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000<div>
<br>
<br><br><?xml version="1.0" encoding="ISO-8859-1" ?><br><!DOCTYPE scenario SYSTEM "sipp.dtd"><br><br><br><scenario name="Basic Sipstone UAC"><br></div> <send retrans="500" start_rtd="1" start_rtd="2"><br>
<br> <![CDATA[<br><br> INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<div><br> Max-Forwards: 70<br> Contact: <sip:[field1]@[local_ip]><br>
From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]<br> To: [service] <sip:[service]@[remote_ip]:[remote_port]><br> Call-ID: [call_id]<br> CSeq: 1 INVITE<br> Content-Type: application/sdp<br>
Content-Length: [len]<br><br> v=0<br> o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br> s=-<br> c=IN IP[media_ip_type] [media_ip]<br> t=0 0<br> m=audio [media_port] RTP/AVP 0<br>
a=rtpmap:0 PCMU/8000<br><br> ]]><br> </send><br><br> <recv response="100"<br></div> optional="true" rtd="1"><br> </recv><br><br><br> <recv response="302" rtd="2"><div>
<br>
</recv><br><br> <send><br> <![CDATA[<br><br> ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br></div><div> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]<br>
</div><div> From: [field1] <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number]<br>
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]<br> Call-ID: [call_id]<br> CSeq: 1 ACK<br> Max-Forwards: 70<br> Content-Length: 0<br><br> ]]><br> </send><br>
<br> <!-- definition of the response time repartition table (unit is ms) --><br> <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/><br><br> <!-- definition of the call length repartition table (unit is ms) --><br>
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/><br><br></scenario><br><br><br><br></div><div><div></div><div><div class="gmail_quote">On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <span dir="ltr"><<a href="mailto:mayamatakeshi@gmail.com" target="_blank">mayamatakeshi@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex"><div>On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<<a href="mailto:mayamatakeshi@gmail.com" target="_blank">mayamatakeshi@gmail.com</a>> wrote:<br>
> On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<<a href="mailto:tculjaga@gmail.com" target="_blank">tculjaga@gmail.com</a>> wrote:<br>
>><br>
>> sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s<br>
>> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l<br>
>> 4000<br>
>> scenario file: uac_redirect.xml<br>
>> FS dialplan: public.xml<br>
>> SIP trace: trace.log<br>
><br>
> The Via definition in your SIPp scenario differs between the INVITE and the ACK:<br>
><br>
> INVITE:<br>
> Via: SIP/2.0/[transport] [local_ip];branch=[branch]<br>
><br>
> ACK:<br>
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]<br>
><br>
><br>
> In the INVITE, you are not adding the [local_port] as you do in the ACK.<br>
> Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK.<br>
> So it seems FS is not checking just the ACK's branch against the<br>
> INVITE's; it seems it is checking the whole Via header.<br>
> I don't know if this is in accordance to SIP specs.<br>
> Another thing, about the way you are calling SIPp: do no use "-sn uac"<br>
> and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"<br>
> means "use the internal (embedded) scenario named xxx". So this<br>
> conflicts with the other parameter "-sf" which specifies an external<br>
> profile.<br>
<br>
</div>I mean, an external scenario (file).<br>
<div><div></div><div><br>
It seems this doesn't cause any problem (probably because in<br>
> the sipp startup, -sf overrides -sn), but it is misleading.<br>
><br>
> regards,<br>
> takeshi<br>
><br>
<br>
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