3 ms·

I'm seriously considering trying something like this just to see if it works. TCP would kill it though (maybe?). But maybe UDP plus some clever ffmpeg invocati

• rvense 3y ago
  I think the main issue would be the buffering existing tools are likely to assume. You don't need more than a few hundred ms of latency for a phone call to start feeling really weird.
• shrubble 3y ago
  If you went with the heavier-weight use of Asterisk, you could have softphones authenticated to Asterisk, but, no inbound calls from the softphones would be accepted. Only if 1 user sought to call another would Asterisk "dial" each softphone; which might be sufficient in terms of security (you would have to have a working SSH account to be called).
• ramses0 3y ago
  https://unix.stackexchange.com/questions/750015/send-alsa-audio-via-network-via-udp https://unix.stackexchange.com/questions/750015/send-alsa-au... https://bbs.archlinux.org/viewtopic.php?id=43896 https://bbs.archlinux.org/viewtopic.php?id=43896 (circa 2008 people were doing this... semi-unironically ;-)