32de076b18
FS-11751: process_rtcp_packet() bounds check
2019-06-06 16:59:08 +00:00
0a02a33023
FS-11480: revert change for FS-11424 which introduced screeching after muxing
...
(thanks Sergey KHripchenko)
2019-06-06 16:56:12 +00:00
2997ceb56f
FS-11678: add/fix muxing to 2 channels (when audio file is multichannel - more than 2 channels)
2019-06-06 16:54:19 +00:00
99bd3b3c6e
FS-11675: RTCP NACK at start of call too
2019-06-06 16:50:45 +00:00
133b42b1e9
FS-11635: decrement RTCP counters on duplicate received NACKed packet which is dropped anyway
...
(affects cum_lost calculation)
2019-06-06 16:50:06 +00:00
a8d7c72084
FS-11666: RTCP: fix RENACK_TIME check (throttle when NACKing the same seq)
2019-06-06 16:49:13 +00:00
bd90f82e6f
FS-11635: RTCP: fix SSRC in FIR, cum_lost due to dup NACKs
...
(thanks Artem Karukov for finding these)
FS-11635: include received Nacked to the expected packet count.
2019-06-06 16:47:29 +00:00
8185aadb5e
FS-11631: [core] RTCP: set RTP timestamp for video in SR (sender info block)
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FS-11631: [core] fix introduced by minor change of behavior, rtcp & ice (FS-11549) -keep rtcp disabled during ICE
2019-06-06 16:42:28 +00:00
f5a736f037
FS-11549: don't send the reception report block when we need to send fast feedback (do like Chrome does it, don't confuse peer's stats).
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FS-11549: don't send RTCP RR unless we're receiver only.
2019-06-06 16:41:43 +00:00
d6169a3845
FS-11561: [core] RTCP: attempt to avoid race cond on calculating DLSR (stats->last_recv_lsr_local)
2019-06-06 16:38:58 +00:00
9682d2c2c9
FS-11730 Add support for DTLSv1.2 and make default
...
Needed in Chrome version >= 74 as Chrome dropped support for v1.0:
https://groups.google.com/forum/?utm_medium=email&utm_source=footer#!msg/discuss-webrtc/yr6gbAgWsKo/9X5vQb8kGAAJ
Using old v1.0 DTLS if 1.2 is not available or legacy DTLS
wanted explicitly. To request old DTLS set variable
<action application="set" data="legacyDTLS=1"/>
Note: requires openssl 1.0.2 or later for DTLS v1.2 support
2019-06-06 12:05:11 -04:00
a095c83ecc
FS-11832 [core] clear the old pmap when codec negotiated
2019-06-06 11:59:16 -04:00
e972499e4f
FS-11775 [core] wait_for_silence debug messages showing as ERR
2019-06-06 11:57:41 -04:00
c786efa9aa
FS-11774: [mod_conference] Quiet down logs from conference moderator command over verto
2019-06-06 11:54:27 -04:00
e47879604c
FS-11824 [mod_sofia] map SIP 607 to SWITCH_CAUSE_CALL_REJECTED
2019-06-06 11:52:07 -04:00
641300893f
FS-11824 [core] add NOT_ACCEPTABLE string to cause mapping
2019-06-06 11:52:01 -04:00
0a6d87edfe
FS-11824 [sofia-sip,mod_sofia,core] Added cause codes that map to SIP 600,603,604,606,607
2019-06-06 11:51:51 -04:00
552d3d8b02
FS-11694 [mod_conference] conference vid-res showing -ERR no reply
2019-06-06 11:50:05 -04:00
372cfc3221
FS-11719 [mod_commands] uuid_broadcast yields OK on bogus uuid
2019-06-06 11:49:22 -04:00
0ba26f0351
FS-11718 [mod_commands] uuid_break yields -ERR no reply
2019-06-06 11:48:16 -04:00
c82f093419
FS-11711: fix overflow on reading ICE candidates (found with ASAN)
2019-06-06 11:47:13 -04:00
90ea0b7a1f
FS-11624: [Build] Fix paralell build of rayo, ssml, digaling can cause iksemel lib build error.
2019-06-06 11:45:21 -04:00
72986c6919
FS-11679 [mod_sofia] SIP message sent by FS logged incompletely in tport dump log
2019-06-06 11:44:27 -04:00
cf7a1d47bf
FS-11640 fix const char * prototype
2019-06-06 11:39:58 -04:00
463eb7294e
FS-11671 fix potential devided by zero error
2019-06-06 11:39:05 -04:00
ab129ffece
FS-11685 [Configuration] Add audio/wav to mime.types config file
2019-06-06 11:37:21 -04:00
60be42008c
FS-11660 [mod_sofia] Not parsing sdp after first video reject.
2019-06-06 11:36:26 -04:00
2192d5de3b
FS-11644: [core] Fix send_dtmf containing w or W
2019-06-06 11:29:21 -04:00
1bd9fd1a86
FS-11785 [sofia] fix dead assignment in parse_payload
2019-06-06 11:25:26 -04:00
8925519e02
FS-11785 [libteletone] Fix dead assignment in teletone_dtmf_detect
2019-06-06 11:25:19 -04:00
7187e8ecd2
FS-11785 [libteletone] Fix unlikely divide by zero in teletone_mux_tones
2019-06-06 11:25:13 -04:00
96c8cb0473
FS-11625: [core]: Do not hangup sessions that have transfer or park after bridge set.
2019-06-06 11:22:37 -04:00
0c73aa82da
FS-11822 [core] reset the codec_string when stream changed
2019-06-06 11:21:19 -04:00
2d231cdff3
FS-11823 [core] clear out event queue when destroying session
2019-06-06 11:20:16 -04:00
20e370aa2a
FS-11573 fix seg when missing conference layout config
2019-06-06 11:18:56 -04:00
a255bd3d62
FS-11668: [verto] Fix polling of secure verto sockets
2019-06-06 11:17:22 -04:00
8dd5373db4
FS-11467: [mod_verto] Properly handle poll() EINTR and separate `poll()==0` / timeout case
2019-06-06 11:17:00 -04:00
1000d1d8c8
FS-11467: [mod_verto] decreased logging level for log lines that are not errors/warnings
2019-06-06 11:16:52 -04:00
6569a44d85
FS-11496: [core] fix crash with four channel sound file
2019-06-06 11:13:06 -04:00
118dd796f3
FS-11421: [core,mod_sofia] Fix rtp_pass_codecs_on_stream_change to process all sdp settings in the re-invite -- add new filter_codecs app and fix some races and negotiation bugs #resolve
2019-06-06 11:08:52 -04:00
c47752c940
FS-11438: [freeswitch-core] recvonly channels left that way on blind transfer in some cases #resolve
2019-06-06 11:05:20 -04:00
699affa1ef
FS-11509: [Build-System] Fix --enable-address-sanitizer configure flag.
...
FS-11509: [Build-System] Fix unimrcp build with address sanitizer.
2019-06-06 11:02:27 -04:00
143323e6ef
FS-11417: [mod_conference] Unbounded memory growth during screen share #resolve
2019-06-06 10:57:45 -04:00
d61126ca82
FS-11814: [mod_sofia] fix regression in fs_path handling on BYE
2019-06-04 15:46:36 -04:00
1aeb90684a
Merge pull request #1739 in FS/freeswitch from verto-min-js to master
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* commit 'f3c52d578b9461015d741eabda2910f360226494':
FS-11874: [Verto-Communicator] Rebuild verto-min.js to reflect the latest changes to js.
2019-06-03 12:08:57 -05:00
f3c52d578b
FS-11874: [Verto-Communicator] Rebuild verto-min.js to reflect the latest changes to js.
2019-06-03 21:00:33 +04:00
f3fdbc5aff
Merge pull request #1738 in FS/freeswitch from ~THEHUNMONKGROUP/freeswitch:bugfix/FS-11874-faulty-logic-for-checking-addition-of-multiple-mediastream-objects to master
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* commit '8fdc4433e5a72b2fdec6455c50f0d5ce67d76271':
Fix FS-11874
2019-05-31 13:22:28 -05:00
8fdc4433e5
Fix FS-11874
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Verto JS libs allow passing an attachStreams array of MediaStream objects, which are then supposed to be added to the peer connection in the order they are supplied in the array.
However, there is a faulty logic check prior to actually adding the streams, where 'options.attachStream' is checked for length instead of 'options.attachStreams'.
2019-05-31 11:07:11 -07:00
ed135a21d5
Merge pull request #1737 in FS/freeswitch from ~THEHUNMONKGROUP/freeswitch:bugfix/FS-11873-addstream-adds-audio-video-tracks-in-random-order-use-spec-compliant-addtrack to master
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* commit 'f1bc56d65b44f1d97e1e9c690175fa8757e42c29':
Fix FS-11873: addStream() adds audio/video tracks in random order
2019-05-31 12:55:51 -05:00
f1bc56d65b
Fix FS-11873: addStream() adds audio/video tracks in random order
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addStream() is a deprecated WebRTC API method, and only works via shimming on some platforms. Internally, it is inconsistent in building an offer SDP in the order which FreeSWITCH expects it, which is audio m-lines first.
This can result in intermittent m-line ordering errors on the client.
Instead addTrack() can be used to force the media tracks to be added to the peer in the order FreeSWITCH expects.
2019-05-31 10:52:18 -07:00
Dragos Oancea
Piotr Gregor
Liyang
joshebosh
Mike Jerris
Chris Rienzo
Seven Du
Mariah Yang
Brian West
Andrey Volk
Sergey Khripchenko
Anthony Minessale
Andrey Volk
Mike Jerris
Chad Phillips