[asterisk-dev] rtp data in an app question

[Gallmeier, Jonathan]

Hi,

 

I'm pretty new to asterisk app development, so pardon me if this
question is old or has been answered elsewhere.

 

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.

 

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?

 

Is there an easy way to programmatically set the RTP packet size in my
application? 

 

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.

 

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. ;)

 

Thanks!

 

Jonathan

 

 

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