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<div class="moz-cite-prefix"><tt>Hi, Marco!<br>
<br>
From RTPProxy point of view, you can't differentiate between SIP
replies, because for all of them you call the same function -
rtpproxy_answer().<br>
Now, if the client decides to send RTP for 183 (and indeed, I've
seen this several times), there's not that much that you can do.
Although it's kind of a hack, all I can think of is to not call
rtpproxy_answer() for 180/183 and strip the body to prevent the
client from sending RTP directly to the callee.<br>
I hope this works for you.<br>
<br>
Best regards,<br>
</tt>
<pre class="moz-signature" cols="72">Răzvan Crainea
OpenSIPS Solutions
<a class="moz-txt-link-abbreviated" href="http://www.opensips-solutions.com">www.opensips-solutions.com</a></pre>
On 01/21/2015 04:07 PM, Marco Hierl wrote:<br>
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<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"
lang="EN-GB">Dear all,<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"
lang="EN-GB"><o:p> </o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"
lang="EN-GB">first of all I need to apologize that I was
not able to find information about this issue although
I’m sure that I’m not the first one complaining!<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"
lang="EN-GB"><o:p> </o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"
lang="EN-GB">The caller is sending an INVITE via
OpenSIPS and rtpproxy_offer() is executed, callee
answers with REPLY 180 or REPLY 183 (with SDP) and
rtpproxy_answer() is made. In this status it should be
ok that the rtp stream from callee to caller is
transferred via the rtpproxy (e.g. for announcements),
but I can see that rtp stream from caller to callee is
transferred too!!! This means that there can be a
conversation without receiving the 200OK and what is the
real problem: that means (at least for me) they can talk
to each other without any charging !! A timer will stop
the conversion after the a while, but this can take
time.<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"
lang="EN-GB"><o:p> </o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"
lang="EN-GB">How can I overcome this problem? How can
prevent RTP to be send to the callee before REPLY 200 is
received?<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"
lang="EN-GB"><o:p> </o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"
lang="EN-GB">I can’t find any help in the RTPproxy
protocol <a moz-do-not-send="true"
href="http://www.b2bua.org/wiki/RTPproxy/Protocol">http://www.b2bua.org/wiki/RTPproxy/Protocol</a>,
nor in the rtpproxy module description in OpenSIPS.<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"
lang="EN-GB"><o:p> </o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"
lang="EN-GB">Thanks for your ideas, and best regards<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"
lang="EN-GB"> Marco<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
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