rst: remove psize, use sampc instead
avcapture: use warning() instead of re_printf()
dshow: use info() instead of re_printf()
rst: use logging functions
modules: add USE_AVCAPTURE and try to detect it
call: check common audio codecs for incoming call

for incoming calls, check that we have at least 1 common audio
codec. if there are no common audio codecs, reject the call
with 488 status code.

Suggested and Tested by Juha Heinanen -- thanks!
ice: set default candidates for 'lite' mode

when using ice-lite mode, the baresip ICE module is not setting
RTP/RTCP connection line in SDP properly. this should be fixed
by making a call to this function after the local candidates
have been gathered:

	icem_lite_set_default_candidates(m->icem);

the issue was reported by Juha Heinanen, thanks
dtls_srtp: clear openssl error queue on errors

openssl has a nice global error queue, if an error occurs
we must read out the error and then call ERR_clear_error().
otherwise other users of openssl in the same process will
get our errors, and things will stop working.
account: add sample with outbound transport
rtp_stats: skip if no packets were sent/recv'd
audio: fix timestamp calculation for sending

according to RFC 3551 the RTP timestamp is independent of
number of channels and encoding. fix the timestamp calculation
for outgoing RTP packets, by dividing the total number of samples
on number of channels.

the bug was not affecting calls with mono audio, but was very
likely to affect calls with stereo audio (e.g. OPUS)
Basic support for X-RTP-Stat reports in BYE/200 OK
fix call mute-state for multiple calls

added audio_ismuted() and remove static state from menu.c

thanks to Remik who reported the issue and suggested
a nice solution :)
Fix return condition to avoid injecting stats when there are none
call: cancel local timer on call_progress()

for incoming calls, we start a local timer of 120 seconds.
this timer is stopped when the call is answered with
200 or 183.

issue was reported by Victor Sergienko, thanks!
move struct stream to core.h
Merge remote-tracking branch 'upstream/master'
Move XRTP stats to audio and implement EN/DE parameters
Minor cleanup
Fixed JI value to timestamp-units based on codec audiorate