[xiph-rtp] speex realtime implementation status

[Kevin Webb]

Hi,

I am working on a internet audio art project and would like to 
incorporate the Speex codec if possible. However, for our application 
we're looking at doing streaming over UDP on a fairly unreliable 
connection and would like to be able to use the RTP payload format - I 
don't thin OGG encapsulation will be ideal.

Is there any implementation out there (even one underdevelopment) that 
implements the RTP spec?

If not do you have any recommendations about how I might be able to make 
Speex work over a low-speed, intermittent UDP connection?

Also just to explain a bit more, we're putting two Mini-ITX boxes on two 
university buses (at Duke University in the US) and using that to create 
a two way audio feed that allows riders to have conversations with folks 
on the other bus... It should be pretty nifty. The only problem is that 
in our area we're limited a Sprint PCS GPRS connection for data service. 
That gives us on average about 70kbps and at worst about 10-20kbps in 
each direction. It's fairly intermittent - which is ok as long as the 
stream can recover from lost packets.

Any advice or ideas would be greatly appreciated. Also we pretty 
comfortable working with your code so if there's anything we can do to 
help with development of the RTP let me know. Our only real issue is 
time - we're hoping to install in April.

Thanks!
Kevin