I need to move realtime audio between two Linux machines, which are both running custom software (of mine) which builds on top of Gstreamer. (The software already has other communication between the machines, over a separate TCP-based protocol - I mention this in case having reliable out-of-band data makes a difference to the solution).
The audio input will be a microphone / line-in on the sending machine, and normal audio output as the sink on the destination; alsasrc and alsasink are the most likely, though for testing I have been using the audiotestsrc instead of a real microphone.
GStreamer offers a multitude of ways to move data round over networks - RTP, RTSP, GDP payloading, UDP and TCP servers, clients and sockets, and so on. There's also many examples on the web of streaming both audio and video - but none of them seem to work for me, in practice; either the destination pipeline fails to negotiate caps, or I hear a single packet and then the pipeline stalls, or the destination pipeline bails out immediately with no data available.
In all cases, I'm testing on the command-line just gst-launch. No compression of the audio data is required - raw audio, or trivial WAV, uLaw or aLaw encoding is fine; what's more important is low-ish latency.
