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<p class=MsoNormal>Hi,<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>I’m pretty new to asterisk app development, so pardon
me if this question is old or has been answered elsewhere.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>I’m writing an asterisk app that bridges a sip call to
a software application that takes and receives RTP audio and video data only
via a named pipe interface. I’ve got the app up and running, but the
audio/video data is not decodable. I’m thinking that I need to handle RTP
packetization across this interface.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Can someone confirm that the data in the asterisk app
channel is sample data? (Asterisk removes the RTP headers before pushing data
into a channel.) If the data is sample data, are are there asterisk RTP
packetization/depacketization APIs that I can use?<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Is there an easy way to programmatically set the RTP packet
size in my application? <o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>While I’m at it, maybe I should ask if this is a common
operation? Am I looking at this problem in the correct way? I’m open to
suggestions on how to best handle this type of bridge operation.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Please point me to useful source code or documentation if
this information has already been covered. I don’t want to rehash basic
information just because I’m new. ;)<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Thanks!<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Jonathan<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
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