gstreamer/gst/rtp

TODO
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- implement packing up to the MTU.
- discont events in the case of packet loss
- figure out the clocking.
- implement various RFCs dealing with different payload types. 
  (as modules?)
- Throw-out the the caps-nego & other session control things to the 
  Application Developer( App ), by turning rtcp work into, signals 
  in gstrtpsend & props/args in gstrtprecv.
  The App would then be free to use any sort of session control 
  protocal like RTSP.( done )


Relevant RFCs
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1889 RTP: A Transport Protocol for Real-Time Applications.

2198 RTP Payload for Redundant Audio Data.
3119 A More Loss-Tolerant RTP Payload Format for MP3 Audio.

2793 RTP Payload for Text Conversation.

2032 RTP Payload Format for H.261 Video Streams.
2190 RTP Payload Format for H.263 Video Streams.
2250 RTP Payload Format for MPEG1/MPEG2 Video.
2343 RTP Payload Format for Bundled MPEG.
2429 RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video
2431 RTP Payload Format for BT.656 Video Encoding.
2435 RTP Payload Format for JPEG-compressed Video.
3016 RTP Payload Format for MPEG-4 Audio/Visual Streams.
3047 RTP Payload Format for ITU-T Recommendation G.722.1.

2733 An RTP Payload Format for Generic Forward Error Correction.
2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony
     Signals.
2862 RTP Payload Format for Real-Time Pointers.
1890 RTP Profile for Audio and Video Conferences with Minimal Control.
2508 Compressing IP/UDP/RTP Headers for Low-Speed Serial Links.


do we care?
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2029 RTP Payload Format of Sun's CellB Video Encoding.

usefull
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http://www.iana.org/assignments/rtp-parameters