651 Commits

Author SHA1 Message Date
Travis Cross
6f11c1636b Add force_send_silence_when_idle channel variable
If set to true, this prevents us from overriding the value of
send_silence_when_idle.  When that is unset or set to zero and SRTP is
engaged, we typically override the value because many devices can't
handle gaps in the SRTP stream.

This variable is mostly for testing whether particular devices can
handle this behavior.  Use at your own risk.
2014-03-04 02:07:30 +00:00
Travis Cross
8fe324c404 Preserve value of send_silence_when_idle if possible
In commit 55d01d3defed4bfdc74704dbea0da9548a97a979 we set
send_silence_when_idle to -1 rather than 400 when SRTP is engaged.
But this left no way to enable white noise silence when desired.

When SRTP is engaged we can't simply not send RTP because it breaks
too many devices.  So we need to prevent send_silence_when_idle from
being unset or being set to zero.  This change allows it to be set to
other values so as to feed white noise rather than all zeros into the
codec.
2014-03-04 02:07:24 +00:00
Anthony Minessale
7ef86923aa FS-5895 --resolve 2014-03-01 04:55:20 +05:00
Travis Cross
b61800cbf2 Send silent packets when idle with SRTP
Originally we did the same thing with SRTP that we do without SRTP,
which is to simply not send packets when e.g. sleep is called.

At commits d63323977fa611b141441f12af9a94ec19b5f829 and
5259814aee16ede974456490a79e8a98de1d6d2e we enabled sending silence
packets with comfort noise when SRTP is active.  We appear to have
done this for interop purposes; many devices can't handle gaps in the
stream of SRTP packets.

But our current comfort noise implementation doesn't take the codec
rate into account (FS-6291), so on 16kHz codecs the constant we chose
created an annoying level of static between sound file playback.

With this commit we preserve the sending of SRTP packets during idle
periods, but make those packets completely silent.

Thanks-to: Anthony Minessale <anthm@freeswitch.org>

FS-5053 --resolve
2014-02-28 23:14:42 +00:00
Anthony Minessale
0c65c4614d FS-6195 --resolve 2014-02-06 22:24:59 +05:00
Anthony Minessale
fa8392a8c0 FS-5675 --resolve 2014-01-24 00:24:15 +05:00
Anthony Minessale
52377f0f65 FS-6007 2013-12-31 00:30:20 +05:00
Jeff Lenk
7712626a02 fix trivial compiler warning for last commit 2013-12-27 14:52:28 -06:00
Anthony Minessale
57184dc20e FS-6007 FS-5083 FS-4514 --resolve 2013-12-28 01:39:03 +05:00
Anthony Minessale
ae6b75a0c2 fix build 2013-11-22 06:08:56 +05:00
Anthony Minessale
69c947f9e8 add polycom support to multicast paging, broadcast all formats at once to support most every popular phone at once
Conflicts:
	src/mod/applications/mod_esf/mod_esf.c
2013-11-22 05:00:26 +05:00
Anthony Minessale
d5d45281f3 change dtmf vol to 13 2013-10-17 21:35:56 +05:00
Anthony Minessale
ab63580e2d FS-5774 --resolve 2013-09-11 03:00:58 +05:00
Anthony Minessale
e1e13a7f5f FS-5754 2013-09-05 03:43:32 +05:00
Anthony Minessale
d816fd9b0e FS-5296 stable version 2013-08-22 02:32:39 +05:00
Seven Du
f9487a9aa8 FS-5670 --resolve allow debug rtp if rtp is not attached to a session 2013-08-08 08:48:40 +08:00
Nathan Neulinger
543dc3caea make noexpandtab explicit in all vim modelines other than xml files
Conflicts:
	src/include/switch_core_media.h
	src/mod/endpoints/mod_sofia/sofia_media.c
	src/switch_core_media.c
2013-06-25 09:39:37 -05:00
Brian West
2cac71675f log auto adjust ip and port separate so it doesn't get clobbered on hold/update events 2013-05-28 16:26:05 -05:00
Travis Cross
4b2dd0a8d9 Don't setup ZRTP without a session and channel
FS-5472 --resolve
2013-05-25 21:46:11 +00:00
Anthony Minessale
3fe3a0091c FS-5418 --resolve 2013-05-15 08:09:07 -05:00
Anthony Minessale
e495702e7d FS-4990 --resolve 2013-04-26 17:23:30 -05:00
Anthony Minessale
63d8df3822 FS-5302 --resolve 2013-04-19 14:07:57 -05:00
Anthony Minessale
3226882682 FS-5203 --resolve 2013-03-27 11:09:07 -05:00
Anthony Minessale
3d2457eddd FS-5120 please update and test this version 2013-03-01 09:29:47 -06:00
Anthony Minessale
b14fd4a5a4 FS-4923 try this 2013-02-14 14:07:25 -06:00
Travis Cross
388206d784 Whitespace 2013-01-31 19:25:07 +00:00
Travis Cross
69cda3d7cc Fix handling of the ZRTP cache path and file name
libzrtp strings are weird, and we were previously misusing the
function zrtp_zstrcpyc.  We can't use ZSTR_GV because it does insane
things and causes an array-bounds warning on gcc 4.7.  So we have to
take matters into our own hands and setup the string correctly and
copy data into it.

Because we were doing it wrong, people would get weird pseudo-random
single-character names for the zrtp cache file, and the file would end
up in the wrong place.  Now that this is fixed, users will need to
locate and move their zrtp cache file to their db_dir and name it
"zrtp.dat" if they wish to keep their current ZRTP cache.

FS-4344 --resolve
2013-01-31 18:40:24 +00:00
Anthony Minessale
a46c897208 use less expensive time func in rtp stack 2013-01-15 09:15:39 -06:00
Anthony Minessale
d78f4ffb19 fix vid regression 2013-01-13 21:27:17 -06:00
Anthony Minessale
ca6d7a7aa0 FS-4905 This looks like it was only the case adjacent to dtmf; Try this patch. 2013-01-08 11:43:16 -06:00
Anthony Minessale
6f376023a2 send cng less often 2013-01-08 11:47:15 -06:00
Brian West
b1ba926640 fix logging in debug def 2012-12-26 10:57:43 -06:00
Anthony Minessale
40346ff6a8 minor tweaks for udptl mode 2012-12-14 16:52:27 -06:00
Jeff Lenk
17833733aa fix for last commit - windows compiler warning 2012-12-13 21:54:54 -06:00
Anthony Minessale
d301451782 fix issue where rtp stack was not paying attn to header extensions 2012-12-13 19:49:02 -06:00
Anthony Minessale
e89fe5a03b FS-4905 hmm, part of my patch did not get comitted... try this... 2012-12-12 09:33:48 -06:00
Anthony Minessale
8aecffef50 don't send cn packet until you have sent some actual audio 2012-12-07 15:11:49 -06:00
Anthony Minessale
924c524197 fix some delay regressions 2012-12-05 13:17:50 -06:00
Anthony Minessale
57a2d59ed6 revert this, probably not needed 2012-12-02 09:58:33 -06:00
Anthony Minessale
444f7924e5 not quite, one more tweak 2012-11-30 14:05:45 -06:00
Anthony Minessale
4e3bb1a1e6 rtp and signal parsing adjustments 2012-11-30 13:54:47 -06:00
Anthony Minessale
6e3c08f590 disable catchup code when jitterbuffer is there 2012-11-30 11:43:48 -06:00
Anthony Minessale
1073f70b6f spread out timer skips to smooth out socket catch-up 2012-11-29 13:43:01 -06:00
Anthony Minessale
d4f8a79299 try to improve audio stream sync 2012-11-28 23:12:35 -06:00
Brian West
89f4cd75f0 Fix 2833 Debuging 2012-11-21 13:59:23 -06:00
Anthony Minessale
6c1e81a813 FS-4797 --resolve 2012-11-06 08:53:00 -06:00
Anthony Minessale
f1bcc387a3 put back seq number when pulling from jb 2012-10-30 16:05:44 -05:00
Anthony Minessale
986fede6b0 make a way to get the jitter buffer down to the codecs 2012-10-18 15:29:00 -04:00
Anthony Minessale
a20aec182a add rtp_manual_video_rtp_bugs 2012-10-16 18:43:00 -04:00
Anthony Minessale
4dd1591945 FS-4684 this should help the proxy situation 2012-10-05 11:49:29 -05:00