Hi,<div><br></div><div>yup i tried &quot;canreinivte=yes&quot; in sip.conf and also in the extension database.</div><div><br></div><div>urm how bout having direct rtp traffic and also relay rtp traffic in my setup?</div><div>
<br></div><div>example, UA1 and UA2 is at the same nat. so UA1 and UA2 will have direct rtp traffic.</div><div><br></div><div>UA1 &lt;--(rtp)--&gt; UA2.</div><div><br></div><div>and UA3 and UA4 both behind different nat will need relay rtp traffic.</div>
<div>when invite compare the &quot;received:ip_address&quot;. </div><div><br></div><div>UA3 &lt;--(rtp)--&gt; KAMAILIO &lt;--(rtp)--&gt; UA4</div><div><br></div><div>issit possible to have both?</div><div>urm ya what is the variable for the &quot;received:ip_address&quot; ? </div>
<div><br clear="all">Thanks.</div><div><br>-- <br>Regards,<br><br>MingHon</div><div><br></div><div><br><div class="gmail_quote">On Thu, Jul 14, 2011 at 11:22 PM, Carsten Bock <span dir="ltr">&lt;<a href="mailto:carsten@ng-voice.com">carsten@ng-voice.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi,<br>
<br>
have you tried &quot;canreinvite=yes&quot; on your Asterisk-box?<br>
If that does not help, there is probably no way to make the<br>
RTP-Traffic bypass your asterisk box...<br>
<br>
Carsten<br>
<br>
<br>
2011/7/14 MingHon &lt;<a href="mailto:gminghon@gmail.com">gminghon@gmail.com</a>&gt;:<br>
<div><div></div><div class="h5">&gt; Hello,<br>
&gt; anyone?<br>
&gt; currently my setup look like this.<br>
&gt; when UA1 call UA2 and the rtp traffic flow to kamailio and asterisk.<br>
&gt; [UA1] &lt;--(rtp)--&gt; [Kamailio/RTPProxy] &lt;--(rtp)--&gt; [UA2]<br>
&gt;                                       ^<br>
&gt;                                       |<br>
&gt;                               RTP TRAFFIC<br>
&gt;                                       |<br>
&gt;                                       v<br>
&gt;                               [ASTERISK]<br>
&gt; what i need to achieve is UA1 call UA2 and rtp traffic flow to kamailio but<br>
&gt; not to asterisk.<br>
&gt; can kamailio handle the rtp traffic it own?<br>
&gt; [UA1] &lt;--(rtp)--&gt; [Kamailio/RTPProxy] &lt;--(rtp)--&gt; [UA2]<br>
&gt;                                      ^<br>
&gt;                                      |<br>
&gt;                                      X<br>
&gt;                                      |<br>
&gt;                                      v<br>
&gt;                              [ASTERISK]<br>
&gt;<br>
&gt; Thanks in advance.<br>
&gt; --<br>
&gt; Regards,<br>
&gt;<br>
&gt; MingHon<br>
&gt;<br>
<br>
<br>
<br>
</div></div>--<br>
Carsten Bock<br>
<a href="http://www.ng-voice.com" target="_blank">http://www.ng-voice.com</a><br>
mailto:<a href="mailto:carsten@ng-voice.com">carsten@ng-voice.com</a><br>
<br>
Schomburgstr. 80<br>
22767 Hamburg<br>
Germany<br>
<br>
Mobile <a href="tel:%2B49%20179%202021244" value="+491792021244">+49 179 2021244</a><br>
Office <a href="tel:%2B49%2040%2034927219" value="+494034927219">+49 40 34927219</a><br>
Fax <a href="tel:%2B49%2040%2034927220" value="+494034927220">+49 40 34927220</a><br>
<br>
~~~~~<br>
Checkout SIP-Provider CE:<br>
<a href="http://www.sipwise.com/products/spce/overview/" target="_blank">http://www.sipwise.com/products/spce/overview/</a><br>
</blockquote></div><br><br>
</div>