Hi Steve,

We've had RTP listed as an experimental feature for a long time, but
it's never been working great. I'm not sure what was wrong, but our
custom use of RTP is either unsound or unsupported by the library
we're using.

If you want low latency, I think there are currently two options. The
first one is the UDP I/O, which can be used with a lossless codec, but
doesn't have any sort of error correction and hence may suffer network
losses --- it is kind of experimental, and was actually created for
artists who were interested in "hearing" such losses, but it may be
useful anyway. Another option is to use jack and a network link using
jack (there was a jack.udp at some point, probably suffering the same
problems as ours) or any other audio library that we support.

Sorry for not being able to provide more solid answers. Keep us
informed of your problems / successes.

Cheers,
-- 
David

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