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&#39;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<br>
<br><br>&lt;?xml version=&quot;1.0&quot; encoding=&quot;ISO-8859-1&quot; ?&gt;<br>&lt;!DOCTYPE scenario SYSTEM &quot;sipp.dtd&quot;&gt;<br><br><br>&lt;scenario name=&quot;Basic Sipstone UAC&quot;&gt;<br>  &lt;send retrans=&quot;500&quot; start_rtd=&quot;1&quot; start_rtd=&quot;2&quot;&gt;<br>
<br>    &lt;![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]<br>      Max-Forwards: 70<br>      Contact: &lt;sip:[field1]@[local_ip]&gt;<br>
      From: [field1] &lt;sip:[field1]@[local_ip]:[local_port]&gt;;tag=[call_number]<br>      To: [service] &lt;sip:[service]@[remote_ip]:[remote_port]&gt;<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>    ]]&gt;<br>  &lt;/send&gt;<br><br>  &lt;recv response=&quot;100&quot;<br>        optional=&quot;true&quot; rtd=&quot;1&quot;&gt;<br>  &lt;/recv&gt;<br><br><br>  &lt;recv response=&quot;302&quot; rtd=&quot;2&quot;&gt;<br>
  &lt;/recv&gt;<br><br>  &lt;send&gt;<br>    &lt;![CDATA[<br><br>      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br>      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]<br>      From: [field1] &lt;sip:[field1]@1[local_ip]:[local_port]&gt;;tag=[call_number]<br>
      To: [service] &lt;sip:[service]@[remote_ip]:[remote_port]&gt;[peer_tag_param]<br>      Call-ID: [call_id]<br>      CSeq: 1 ACK<br>      Max-Forwards: 70<br>      Content-Length: 0<br><br>    ]]&gt;<br>  &lt;/send&gt;<br>
<br>  &lt;!-- definition of the response time repartition table (unit is ms)   --&gt;<br>  &lt;ResponseTimeRepartition value=&quot;10, 20, 30, 40, 50, 100, 150, 200&quot;/&gt;<br><br>  &lt;!-- definition of the call length repartition table (unit is ms)     --&gt;<br>
  &lt;CallLengthRepartition value=&quot;10, 50, 100, 500, 1000, 5000, 10000&quot;/&gt;<br><br>&lt;/scenario&gt;<br><br><br><br><div class="gmail_quote">On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <span dir="ltr">&lt;<a href="mailto:mayamatakeshi@gmail.com">mayamatakeshi@gmail.com</a>&gt;</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 class="im">On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi&lt;<a href="mailto:mayamatakeshi@gmail.com">mayamatakeshi@gmail.com</a>&gt; wrote:<br>

&gt; On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga&lt;<a href="mailto:tculjaga@gmail.com">tculjaga@gmail.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s<br>
&gt;&gt; 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l<br>
&gt;&gt; 4000<br>
&gt;&gt; scenario file:      uac_redirect.xml<br>
&gt;&gt; FS dialplan:       public.xml<br>
&gt;&gt; SIP trace:          trace.log<br>
&gt;<br>
&gt; The Via definition in your SIPp scenario differs between the INVITE and the ACK:<br>
&gt;<br>
&gt; INVITE:<br>
&gt; Via: SIP/2.0/[transport] [local_ip];branch=[branch]<br>
&gt;<br>
&gt; ACK:<br>
&gt; Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]<br>
&gt;<br>
&gt;<br>
&gt; In the INVITE, you are not adding the [local_port] as you do in the ACK.<br>
&gt; Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK.<br>
&gt; So it seems FS is not checking just the ACK&#39;s branch against the<br>
&gt; INVITE&#39;s; it seems it is checking the whole Via header.<br>
&gt; I don&#39;t know if this is in accordance to SIP specs.<br>
&gt; Another thing, about the way you are calling SIPp: do no use &quot;-sn uac&quot;<br>
&gt; and &quot;-sf uac_redirect.xml&quot; at the same time. The parameter &quot;-sn xxx&quot;<br>
&gt; means &quot;use the internal (embedded) scenario named xxx&quot;. So this<br>
&gt; conflicts with the other parameter &quot;-sf&quot; which specifies an external<br>
&gt; profile.<br>
<br>
</div>I mean, an external scenario (file).<br>
<div><div></div><div class="h5"><br>
 It seems this doesn&#39;t cause any problem (probably because in<br>
&gt; the sipp startup, -sf overrides -sn), but it is misleading.<br>
&gt;<br>
&gt; regards,<br>
&gt; takeshi<br>
&gt;<br>
<br>
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