0db9843ea8
FS-8716 actually fix typo
2016-01-08 09:57:52 -06:00
42ac717d58
FS-8716 typo
2016-01-07 16:46:30 -06:00
0da5d8a350
FS-8716 #resolve [recording offset is delayed few seconds for rtmp stream]
2016-01-07 16:39:06 -06:00
56a68e3ad9
FS-8677 #resolve [Crash (possible memory corruption) after codec change]
2016-01-06 10:10:14 -06:00
2cb7188c17
FS-8632 #resolve [Add origination_audio_mode originate variable]
2015-12-08 18:30:48 -06:00
8a73ecc14a
FS-8053 amendment
2015-11-18 12:33:42 -06:00
771123eb1b
FS-8053 typo2
2015-11-18 10:55:05 -06:00
07b8458497
FS-8053 typo
2015-11-18 10:50:40 -06:00
e95f3686ad
FS-8053 #resolve [When WebRTC's SDP contains a=sendonly for video, the client will still receive the video stream]
2015-11-18 10:31:30 -06:00
67491b102c
FS-8543 #resolve [Improve mute handling on conference and WebRTC]
2015-11-16 11:31:34 -06:00
aa0b28e235
WIP not shabby auto vid settings
2015-11-06 14:01:54 -06:00
7d57063a14
FS-8404: if media engine will default to PCMU/PCMA if you don't specify any codecs
2015-11-05 11:30:02 -06:00
38ff603766
FS-8417 #resolve [SIP offer with a=sendonly sometimes replies with a=inactive]
2015-11-05 10:43:19 -06:00
238722879b
FS-8414 #resolve [Ptime unchanged on codec renegotiation]
2015-11-04 09:46:36 -06:00
e96c2d4917
FS-8411 Replace ping_frame with video_ping_frame in a couple places that were missed
2015-11-03 09:27:21 -06:00
e299489cfc
FS-8391 #resolve [SDP parsing error for rtcp-fb]
2015-10-27 10:58:04 -05:00
f6427a5f92
FS-8382 #resolve [Segfault with inbound-proxy-media enabled]
2015-10-22 15:02:38 -05:00
116c4aa916
up default max jb size to 50
2015-10-21 11:50:14 -05:00
847fe0f476
Merge pull request #535 in FS/freeswitch from ~COREY BURKE/freeswitch:bugfix/FS-8286-improve-debug-logging-for-info-level to master
...
* commit '3a9e7f08b4757aeaa33537658379f3964104cb7d':
FS-8286: Minor debug log level tweaks
2015-10-21 11:43:18 -05:00
072f269ee7
FS-8372 #resolve [Wrong response to RTP/SAVPF without DTLS]
2015-10-21 11:28:37 -05:00
3184180238
FS-8368 #resolve [Reduce logging for audio/video sync]
2015-10-20 13:39:10 -05:00
7572b52815
FS-8275 #resolve [RFC2833 DTMF broken in recent master] REGRESSION FIXED
2015-10-20 12:06:40 -05:00
56d83d1967
FS-8321 #resolve [BEHAVIOR CHANGE Add variable media_mix_inbound_outbound_codecs to mix inbound and outbound codecs]
2015-10-08 13:17:24 -05:00
3e23ff7d29
uncomment code
2015-10-07 14:38:36 -05:00
444f81ce9f
FS-8316 fixed new build warning from latest clang
2015-10-07 10:35:46 -07:00
b205355c24
Merge pull request #505 in FS/freeswitch from ~SAFAROV/freeswitch2:fix_udptl_SDP to master
...
* commit '62d4d47b1a09075acc19cd792a90468d50e6cb14':
FS-8198: Fixed default CRLF sequence in t38 SDP
2015-10-07 12:32:53 -05:00
3a9e7f08b4
FS-8286: Minor debug log level tweaks
...
Adjust some DEBUG and INFO log lines, reducing log verbosity at the INFO level while increasing call debugging info.
2015-10-02 08:41:41 -07:00
c71b0cbd86
FS-8275 #resolve [RFC2833 DTMF broken in recent master]
2015-10-01 19:43:51 -05:00
f1c61f6f24
FS-8216 #comment please test
2015-09-30 22:03:33 -05:00
beb1521e88
revert
2015-09-30 22:02:50 -05:00
8401e716b1
FS-8240 #comment please test
2015-09-30 22:00:47 -05:00
d335fb089e
FS-8220 #resolve [DTMF not working between telephone-event/48000 A leg and telephone-event/8000 B leg]
2015-09-28 13:26:07 -05:00
5a18d02e66
FS-8166 #resolve [Mute/unmute while shout is playing audio fails because the channel "has a media bug, hard mute not allowed"]
2015-09-28 12:38:58 -05:00
8b7465425d
FS-8130 regression in bridged channels with jitterbuffers
2015-09-23 14:46:47 -05:00
62d4d47b1a
FS-8198: Fixed default CRLF sequence in t38 SDP
2015-09-22 16:58:08 +03:00
13add97426
FS-8130 refactor pass
2015-09-21 15:32:38 -05:00
8a0366b999
FS-8130 running out of witty commit msgs
2015-09-21 14:08:38 -05:00
0fa190093e
FS-8130 the bug that keeps on giving
2015-09-18 20:54:42 -05:00
4a56b7d145
FS-8130 cont
2015-09-17 17:34:58 -05:00
944bad6708
FS-8173 fix SAVPF printing when it's really AVPF
2015-09-16 18:08:44 -05:00
703795e913
FS-8130 FS-7432 FS-8115
2015-09-15 20:33:55 -05:00
52afedf23d
FS-8053 addtl touchups
2015-09-15 12:14:09 -05:00
36cd926981
typo
2015-09-14 20:04:17 -05:00
b2f97336e8
FS-8114 #resolve [Opus and telephone event payload types collide on REFER]
2015-09-14 19:44:48 -05:00
79f158911e
FS-8053 fix some regressions from original merge, add auto mute-unmute when toggling video send/recv
2015-09-14 16:49:55 -05:00
eea76c8856
FS-8130
...
Port video buffer to also support audio and remove original STFU jitter buffer
Add some more resilience to video packet loss
Add codec control mechanism for both call-specific debug and codec/call specfic params
Make opus function better in packet loss and latent situations
Use new codec control prams to make JB lookahead FEC optionally enabled or disabled mid-call
Add Param to allow JB lookahead to be enabled.
2015-09-14 13:30:08 -05:00
331cde3c92
FS-8053: handle a=sendonly, a=sendrecv, a=recvonly to change who is sending video during a call
2015-09-09 18:10:52 -05:00
fe0908d6e1
FS-8126 #resolve [Pruning of a media bug may cause all media bugs on a session to be pruned]
2015-09-09 13:04:58 -05:00
293ceabb78
let jb get bigger on video sync
2015-09-04 13:53:50 -05:00
8ee69ead0d
FS-8118 #resolve [Verto calls do not properly reject video when video is offered but only audio is accepted]
2015-09-02 18:07:58 -05:00
Anthony Minessale