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<p>Dear Asterisk-Users,</p>
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<p>By means of Asterisk 11 and sip.conf, I got success implementing early media. That is, all information that come from callee (SIP 183 message/ SDP) is passed to the caller without any modification in the SDP body.</p>
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<p>However, in Asterisk 13 and using pjsip.conf I'm still failing to do the same thing. See:
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<p>Softphojne1 <------------------------------------------------> Asterisk <--------------------------------------->Softphone2</p>
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<p> | ----------------------SIP INVITE ------------------------>|</p>
<p> | ----------------------------SIP INVITE------------>|</p>
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<p> . .</p>
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<p> |<----------SIP 183 ---------------------------------|</p>
<p> SDP : Media Description, name and address (m): audio 4000 RTP/AVP 8 96</p>
<p> Media Description, name and address (m): video 5000 RTP/AVP 97<br>
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<p> |<--------------------------SIP 183-----------------------------|</p>
<p> SDP :Media Description, name and address (m): audio 13258 RTP/AVP 8 96
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<p> Media Description, name and address (m): video 16002 RTP/AVP 97<br>
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<p>So, there is bit modification in SDP body, caused by Asterisk. As long as I'm intending to implement direct media, I believe that Asterisk 13 has some special configuration to be done in PJSIP.conf file, that will allow things work very well again, as in
Asterisk 11 and sip.conf.</p>
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<p>How to configure pjsip.con file or Asterisk, to run direct-media? Or , where to find a tutorial about it on Internet?</p>
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<p>Any hint will be very helpful!</p>
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<p>Thanks a lot.</p>
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<div class="PlainText">RODRIGO PIMENTA CARVALHO<br>
Inatel Competence Center<br>
Software<br>
Ph: +55 35 3471 9200 RAMAl 979 (Brasil)</div>
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