[gst-devel] RTP plugin implementation

[Angel Carpintero]

 Hi all , 

  I've been looking at current rtp plugin implementation and i'm a bit confused.
  The current way of rtp implementation is to build a modul for each payload, i.e. :

   GSM  payload = 3 
   gstrtpgsmenc :  Encodes GSM audio into an RTP packets.
   gstrtpgsmparse :  Extracts GSM audio from RTP packets.

  So it means for each payload should be implement his own *encoder*/*parser* (!!) , right ?
  
  Then a scheme migth be ( for GSM i.e. ) :

   raw-audio <---> gsmenc <--->  gstrtpgsmenc (!!)  <-->  gstrtpsend   

   ....  gstrtprecv <---> gstrtpgsmparse (!!) <--> gsmdec <--> esdsink
   

  About the rtcp , does should it be implemented into gstrtpsend & gstrtprecv , right ?

  Well i will go futher as soon that will be totally clear, then rtp TODO can be fixed.
   
  Please feel free to send your comments & opinions . 

  Thanks !

  
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-- 
Angel Carpintero - angel at e-group.org
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