- Add apt-get install lines for Ubuntu 18.04
- add gstreamer-webrtc-1.0 and gstreamer-sdp-1.0 to CFLAGS
- make the CLAGS match LIBS in Makefile dependencies
We don't own the reference. Since GLib 2.58, the g_bytes_unref that
follows the signal emission in libsoup loudly complains about the
attempt to underflow the refcount.
When using localhost signalling servers, we don't want to use
strict ssl because it's probably using a self-signed certificate
and there's no need to do certificate checking over localhost anyway.
Missed this, fixes the bug where removing a peer causes the pipeline to
get stuck. However, when peers leave, there is still a chance that the
pipeline will get stuck.
You can join a room and an audio-only call will be started with all
peers in that room. Currently uses audiotestsrc only.
BUG: With >2 peers in a call, if a peer leaves, the pipeline stops
outputting data from the remaining peers to the (audio) sink.
TODO: JS code to allow a browser to join the call
TODO: Cleanup pipeline when a peer leaves
TODO: Add ICE servers to allow calls over the Internet
TODO: Perhaps setup a TURN server as well