this project is about implmenting the rtp layer and codec layer of [url removed, login to view] codec.
the library should not handle jitter buffer/network/codec.
the interface will be callback to get udp packet which will be parsed by the library and return to the next layer the packets with the timestamp for them.
on the other hand, it will get compressed packet from the codec, pack them to rtp packet and pass them to the next layer (network).
* * *This broadcast message was sent to all bidders on Tuesday Nov 24, 2009 2:22:41 AM:
based on some quastion i got: 1. I need one RTP channel. 2. the interface from the network layer is by 2 functions. one to recieve buffer and one to send buffer - the rtp doesn't handle IP/port, just send and recieve. 3. on the TX part there should be 2 function to send, one to add audio packet and one to flush to the network (pack several audio frame into one rtp packet). 4. on the RX part, the rtp layer just suppose to parse the packet to its audio frames and pass them to the jitter buffer with the corect sequence number (play time) for each frame. 5. no RTCP channel! 6. no use of extension of the RTP header, just 12 byte. 7. you may use open source but I need very very light RTP layer so I prefer not using open source as they are over kill for this mission.