Hello, I'm trying to work out how to control latency with pulseaudio CLI scripts. We're finding that latency varies between a few seconds to about 80 seconds. We have a system which uses a dedicated embedded board for many channels of audio I/O. Workstations connect with the I/O board using RTP over a network. Pulseaudio 8.0 is used on both I/O board and workstation platforms, both using pulseaudio CLI scripts. Modules explicitly loaded include instances of module-rtp-send, module-rtp-recv, module-null-sink, module-remap-source, module-remap-sink and module-loopback. This all works as far as it goes, but with VoIP (using 1 channel in each direction), we're finding that the latencies make it pretty much unusable. Ideally, I need to be able to put a reasonable upper limit on total latency. The link https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/Developer/Clients/LatencyControl/ provides instructions for use with the API, but I can't find much about controlling latency with CLI. A few modules appear to have latency-related parameters I can tweak, but this seems to be pointless because other modules are adding latency that I haven't worked out how to control. Is there any way to do this? Steve