f38dde9fec
fix merge
2016-09-27 16:44:25 -05:00
c409499cd9
FS-9576 #resolve [Add Realtime Text]
2016-09-27 16:40:43 -05:00
6d6bd1efa5
FS-9242 convert to adapter.js
2016-09-27 16:07:00 -05:00
0e6e53f15c
FS-8644: OPUS_SET_BITRATE(), codec control and estimators for packet loss and RTT (with Kalman filters) to detect a slow or congested link.
...
Feature enabled with "adjust-bitrate" in opus.conf.xml - it's a feedback loop with incoming RTCP.
2016-09-27 16:06:17 -05:00
9edcbfe036
FS-9522 more regression
2016-09-26 20:06:30 -05:00
2b1f0da5c4
FS-9525 part 2
2016-09-23 16:50:47 -04:00
25732c82d2
FS-9557
2016-09-22 19:52:05 -05:00
eb09fb36eb
FS-9557 #resolve [Eating AV in proxy media mode]
2016-09-22 18:52:48 -05:00
d34b501758
FS-9435 #resolve [PLI requests once per second]
2016-09-20 17:18:10 -05:00
2bd7cfdf9a
FS-9522 #resolve [Add rtp_assume_rtcp to always use rtcp when needed]
2016-09-16 12:01:43 -05:00
0af2659691
FS-9511 up the max size a tad
2016-09-15 15:49:47 -05:00
8eda172aab
FS-9503 #resolve [Add flaws and consecutive_flaws to error_log in rtp]
2016-09-12 12:21:21 -05:00
f3b22f6142
FS-9496
2016-09-12 12:04:21 -05:00
2636141c32
Revert FS-8565, its getting a few blames for 3cx interop issues, we need to revisit the original reason and make sure this was the right fix.
2016-08-18 15:12:28 -05:00
b9733c577c
FS-9436 #resolve [RTCP PLI Media Source SSRC wrong after re-INVITE]
2016-08-17 18:22:20 -05:00
eeba18394d
FS-9410 #resolve [PLI Missing Media Source SSRC]
2016-08-12 14:28:45 -05:00
f34dac749e
revert FS-9368
2016-07-28 14:13:37 -05:00
9879731f8f
FS-9334
2016-07-21 22:11:29 -05:00
c556e5269b
FS-9368
2016-07-21 14:32:04 -05:00
ec2734c10c
FS-9334 revert
2016-07-21 03:22:37 -05:00
d4daa8a051
FS-9246 #resolve [No Audio after Call transfer ]
2016-06-10 15:01:37 -05:00
23a1d37ae7
FS-9131: improve validation of ice candidates
2016-05-04 13:34:40 -04:00
aadf89620a
FS-8949 #resolve [switch_rtp.c Send end packet for [X] not recognised]
2016-04-15 19:36:47 -05:00
29263bf7b9
FS-8623: fix sun studio error trying to compile char[] with c++ compiler
2016-04-07 15:11:42 -04:00
fe6182260b
FS-9013 #resolve [Add vad talk time logging channel vars]
2016-03-31 19:25:56 -05:00
e18c12b609
FS-8953 [core] white space clean up.
2016-03-17 08:55:00 -05:00
0b8fbb47b2
FS-8891 #resolve [T38 fax fails between 2 freeswitch boxes, with high cpu usage (comparison overflow?)]
2016-03-03 09:32:12 -06:00
339cef5d05
FS-8864
2016-02-26 17:44:51 -06:00
e4e752fdde
FS-8864
2016-02-26 17:42:41 -06:00
6334e1ce39
FS-8864 #resolve [Improve video file playback]
2016-02-26 15:28:03 -06:00
3bd26eaa6b
FS-8847 #resolve [Silence Error on shutdown of video call]
2016-02-19 09:34:38 -06:00
60bd4bec38
FS-8802 #resolve [RTP stops sending audio when sent timestamp rolls over]
2016-02-01 16:23:43 -06:00
a729ebe6e5
FS-8786 #resolve [rtp-timeout-sec on profile triggers when receiving T.38 UDPTL frames]
2016-01-27 10:49:44 -06:00
013fb1d29c
Merge pull request #652 in FS/freeswitch from ~SJTHOMASON/freeswitch:bugfix/FS-8571-configure-with-disable-srtp-breaks to master
...
* commit 'e3e6bd875747c379ae78e7528dc66ce210c24ed9':
Add missing ENABLE_SRTP ifdef to allow building without SRTP
2016-01-21 23:32:49 -06:00
5de4622734
Revert "FS-8713 #resolve avoid read exceeding buffer"
...
This reverts commit 1746d2ebc0
2016-01-07 17:00:48 +08:00
1746d2ebc0
FS-8713 #resolve avoid read exceeding buffer
2016-01-07 14:11:48 +08:00
56a68e3ad9
FS-8677 #resolve [Crash (possible memory corruption) after codec change]
2016-01-06 10:10:14 -06:00
e3e6bd8757
Add missing ENABLE_SRTP ifdef to allow building without SRTP
...
FS-8571 #resolve
2015-12-29 15:50:38 +00:00
1d36d1fcae
Merge pull request #606 in FS/freeswitch from ~MBRANCA/freeswitch:bugfix/FS-8565-txfax-fails-when-remote-party-changes to master
...
We've tested this with Yehavi and it does fix a problem that apparently happens a lot in the wild, and I have tested it against FS.
* commit 'd7b8394e501ae061926a78a1d45a0be8aafb6907':
FS-8565 update initial rtp from address when auto changing port so we can avoid false later comparisons
2015-12-08 09:58:16 -06:00
03a9c7e11c
FS-8575 #resolve [No DTMF passed from a to b during rfc 2833 events.]
2015-11-25 14:45:32 -06:00
e31a8999c7
Allow building using system OpenSSL without EC support
...
This allows FreeSWITCH to build using system OpenSSL when there is no EC support
FS-8568 #resolve
2015-11-24 19:44:49 +00:00
d7b8394e50
FS-8565 update initial rtp from address when auto changing port so we can avoid false later comparisons
2015-11-23 12:24:29 +01:00
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
860a799c1d
FS-8547 #resolve [Add error log into stats to log when quality impacting events begin and end]
2015-11-17 15:09:00 -06:00
cb7065f7b0
Merge pull request #594 in FS/freeswitch from bugfix/FS-8425-dtmf-sometimes-missed-on-pstn-call to master
...
* commit '56a10538dc6307fb285dd1f3a31b384f0e5750fa':
FS-8425 #resolve [DTMF sometimes missed on PSTN call]
2015-11-13 10:56:08 -06:00
56a10538dc
FS-8425 #resolve [DTMF sometimes missed on PSTN call]
2015-11-13 10:55:47 -06:00
cc1bf31078
FS-8534: calculate RTT average (RTCP SR)
2015-11-12 18:50:38 -05:00
9edede6c08
Merge pull request #546 in FS/freeswitch from ~COREY BURKE/freeswitch:feature/FS-8281-expose-srtp-and-srtcp-crypto-keys to master
...
* commit '0316fdfcf1d2a72d1bad5dc9646371159173f93f':
FS-8281: Expose SRTP and SRTCP crypto keys as channel vars
2015-10-27 19:13:25 -05:00
0316fdfcf1
FS-8281: Expose SRTP and SRTCP crypto keys as channel vars
...
New vars are srtp_{local,remote}_crypto_key and srtcp_{local,remote}_crypto_key.
Allows decrypting packet captured media streams for debugging.
2015-10-27 16:33:49 -07:00
7e47db6b9b
FS-8130 regression causing excessive mark bit detection in some cases
2015-10-22 13:07:18 -05:00
Anthony Minessale
Dragos Oancea
Brian West
Michael Jerris
Ken Rice
Anthony Minessale II
Seven Du
Spencer Thomason
matteo brancaleoni
Corey Burke