BigBlueButton/bigbluebutton-Github
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220 Commits

Author SHA1 Message Date
Richard Alam
75abfce102 - upgrade bbb-video and bbb-voice to red5 1.0 rc1 2011-05-04 11:03:57 -04:00
Richard Alam
14dc18e392 - add util class to dump bytes to a file...useful for debugging 2011-04-04 11:04:57 -04:00
Richard Alam
775d7fff3c - cleanup 2011-03-22 10:22:44 -04:00
Richard Alam
2e821681f0 - add check for logging 2011-03-02 01:34:15 +00:00
Richard Alam
e7a3fa690e - fix start/stop stream errors 2011-03-02 01:27:01 +00:00
Richard Alam
15459681f6 - minor cleanup and add start/stop into transcoder 
TODO:
  - cleanup logging
  - create abstract class to implement common methods that individual transcoders override
 2011-03-01 21:42:42 +00:00
yuan
a86c7e6186 Support to fix speex wideband 2011-02-28 14:49:40 +00:00
Fred Dixon
20058eb495 - Assigned values for startAudioPort=15000 and stopAudioPort=16383 in bigbluebutton-sip.properties 2011-01-10 21:12:32 -05:00
Fred Dixon
cd5c510df3 - Change value for startAudioPort and stopAudioPort to be 16384 and 32767 2011-01-10 19:19:03 -05:00
Richard Alam
a246f5452c - modify to use NIO buffer and add more documentation to transcoding process 2011-01-04 16:20:28 -05:00
Richard Alam
5941e6371e - drop packets when there is connection congestion 2011-01-04 11:47:40 -05:00
Richard Alam
bd7c7bd17f - send two of the remaining 3 packets at the same time to minimize choppy audio if we just dropped the 3 extra packets 2011-01-03 13:48:07 -05:00
Richard Alam
e49e1cc83b - use FloatBuffer to store transcoded audio 2011-01-03 12:50:06 -05:00
Richard Alam
6151a60cfc - cleanup 2010-12-15 17:05:01 -05:00
Richard Alam
71588b22de - change how we increment timestamps for audio packet 2010-12-14 15:38:09 -05:00
Richard Alam
bcd8d07b9f - change how we put timestamps into the audio packet and mark the packet as live 
that way the RTMPProtocolEncoder can filter packets and start dropping those
   that have been in the queue for long
 2010-12-07 16:09:27 -05:00
Richard Alam
a9c7605fad - change audio packet queues into pipedinput/outputstream 
- drop audio bytes if it grown larger than 1000
 2010-12-07 13:51:01 -05:00
Richard Alam
19066eb91a - null rtppacket to check garbage collection issues 2010-12-03 19:14:23 -05:00
Richard Alam
02970d2b4d Merge branch 'master' of github.com:bigbluebutton/bigbluebutton 2010-12-03 18:53:06 -05:00
Richard Alam
a01541ec1c - send and receive udp packets only from the specified address 2010-12-03 18:50:50 -05:00
Scott Morris
3261a669a2 Updated some debug issues 2010-12-03 18:05:39 -05:00
Richard Alam
68f94062c1 - merge scott's fixes for voip threads 
Conflicts:
	bbb-voice/src/main/java/org/bigbluebutton/voiceconf/red5/media/SipToFlashAudioStream.java
 2010-12-02 16:07:44 -05:00
Scott Morris
e0a0e510e8 Added posioning support the the audioByteData class and check to see if a posioned packet has been added to the queue. If so then stop consuming packets from the queue. This fixes the left over audio threads. 2010-12-02 12:45:53 -05:00
Richard Alam
e5f1536ae2 - add more logging on why local UDP port for audio is hanging around 2010-12-01 12:12:44 -05:00
Richard Alam
75b6f10582 - increase delay check time as we notice we are dropping too many packets when Asterisk/FreeSWITCH is on a different server 2010-11-29 10:24:42 -05:00
Richard Alam
9ba6b2e878 - add logging when FS/Asterisk is the one telling us to hangup (e.g. being kicked from the conference) 2010-11-26 14:27:06 -05:00
Richard Alam
4665abd490 - change the logs so it's a little bit clear on what the user is doing 2010-11-26 13:46:17 -05:00
Richard Alam
db8ba6cd14 - fix log format 2010-11-25 16:32:49 -05:00
Richard Alam
095f532e35 - add more info on log so we can correlate with the red5 error.log if the client dropped because of connection problems 2010-11-25 15:59:07 -05:00
Richard Alam
b456c92822 - add meaningfull logging so we can track a user when joining/leaving conference. 2010-11-25 15:38:38 -05:00
Richard Alam
c28258d1f3 - format log a little bit better so as not to flood logging when we go thourhg a lot of ports and fail 2010-11-23 17:02:29 -05:00
Richard Alam
98e950ce13 - add more debugging info 2010-11-23 14:51:19 -05:00
Richard Alam
55a10e750a - aggressively try to get a local audio port 2010-11-23 14:42:39 -05:00
Richard Alam
480c3e990d - change license headers for bbb-voice 2010-11-06 11:30:32 -04:00
Richard Alam
c8aa90c790 - cleanup 2010-10-27 14:32:17 -04:00
Richard Alam
d3bc9fd29a - fix problem where audio is silent because of how we set the fake metadata timestamp 2010-10-05 12:25:47 -04:00
Richard Alam
17b3f3bdae - cleanup and add comment on possible reason why Asterisk sends RTCP 2010-09-25 11:14:08 -04:00
Richard Alam
00b2759bfd - add comments and fix timestamps (should be incremented based on codec not based on clock) 2010-09-24 10:48:37 -04:00
Richard Alam
a44648a515 - drop delayed RTP packets 
- add some comments
 2010-09-24 10:47:47 -04:00
Richard Alam
b5b427298a - remove debug logs 2010-09-24 10:46:52 -04:00
Richard Alam
7f29dfe3b0 - add some debug logs to determine how long Red5 is receiveing audio packets from the client 2010-09-24 10:43:10 -04:00
Richard Alam
1f9457395a - make debug logging only when debug id enabled 2010-09-21 14:25:29 -04:00
Richard Alam
54fa14a809 - handle (discard) RTCP packets properly 2010-09-21 12:21:22 -04:00
Richard Alam
2ab97b4727 - just process received rtp packet without trying to figure out the sequence number to drop 
delayed packets.
 2010-09-20 15:25:27 -04:00
Richard Alam
fd1a87bea9 - handling the 52 byte packet doesn't work quite well...we keep on hearing a clicking sound. 
Reverting to the old way of throwing away incorrect size packets.
 2010-09-20 15:23:03 -04:00
Richard Alam
59023f6304 - see what happens when handling the 52 byte ulaw packet from Asterisk 2010-09-20 15:19:31 -04:00
Richard Alam
1134104118 - add log to display incoming frame length when not equal to expected frame length. Need to figure out 
why we are getting incorrect frame length under heavy load
 2010-09-20 11:11:06 -04:00
Richard Alam
e05555f2ad - if we drop 3 consecutive rtp audio packets, reset the stream to handle the next incoming packets. 
This way, we avoid dropping all remaining packets resulting in a silent stream for the unfortunate user.
 2010-09-20 10:45:41 -04:00
Richard Alam
2018c74a81 - recognize market rtppacket to handle changes in rtp seq num and timestamp in the middle of the stream 2010-09-19 12:57:57 -04:00
Richard Alam
8b171a6055 - clean up some more 2010-09-10 10:46:40 -04:00
Richard Alam
3bed51ca1e - minor cleanup 2010-09-10 10:37:58 -04:00
Richard Alam
42433ffccb - handle rolling over of sequence number. Sequence is only 16-bits (65535) and can start at any number so it 
can rollover if the audio stream is too long.
 2010-09-09 16:15:28 -04:00
Richard Alam
6e961caab8 - change log to debug 2010-09-09 14:32:47 -04:00
Richard Alam
eef0c12091 - init last seq num and last timestamp from first packet received. Don't assume that seq num always start at 0. 2010-09-09 13:33:32 -04:00
Richard Alam
5100fff24b - investigate why a user looses incoming audio (can't hear but can still talk) 2010-09-09 11:29:43 -04:00
Richard Alam
1fe65660fd - throw away delayed rtp packets 2010-09-07 15:08:01 -04:00
Richard Alam
bdfd342159 - add comment on where the timestamp values came from so that others won't wonder 2010-08-27 11:58:40 -04:00
Richard Alam
78ca8120c8 - add a recording stream hook...used it to record ulaw and speex stream flv 2010-08-26 13:10:51 -04:00
Richard Alam
8aaccc024c - cleanup of printlns 2010-08-24 15:48:11 -04:00
Richard Alam
b397c6130b - increment timestamps 
Ulaw: 160 (RTMP -> RTP)
           32 (RTP -> RTMP)
    Speex WB: 320 (RTMP -> RTP)
               20 (RTP -> RTMP)
 2010-08-24 15:39:55 -04:00
Richard Alam
905e47e8a7 - generate fake metadata to fix problem when upgrading from FP 10.0 to 10.1 2010-08-24 15:38:59 -04:00
Richard Alam
0d29c05b7f - cleanup 
- change start of timestamp from 0 to a random number from 0-1000
 2010-08-23 13:22:03 -04:00
Richard Alam
7a87fc2a53 - change timestamp for ulaw to 180 increments 2010-08-23 12:04:17 -04:00
Richard Alam
b574413d1e - cleanup and change timestamp for speex flash to sip to 320ms 2010-08-22 10:12:57 -04:00
Richard Alam
bbd6d1904b - cleanup and change speex transcoder to increment timestamp by 20ms. 2010-08-22 10:08:16 -04:00
Richard Alam
3fa480baaf - use refactored RtpPacket 2010-08-20 13:27:44 -04:00
Richard Alam
32019cd622 - modify to use new gradle task 2010-08-20 11:03:32 -04:00
Richard Alam
e0844956b0 - add testng.xml for unit testing 2010-08-20 11:02:57 -04:00
Richard Alam
3edc00369e - refactor RtpPacket and add unit tests 2010-08-20 11:01:20 -04:00
Richard Alam
d820c400c4 - add testng and easymock dependencies 2010-08-20 11:00:09 -04:00
Richard Alam
d08a1bdfdf - set timestamps to increment by 20ms 2010-08-17 16:44:59 -04:00
Richard Alam
db8d73c6a4 - use one thread to process rtp packets 2010-08-17 15:05:06 -04:00
Richard Alam
9faca38368 - display inter-packet arrival time from FS 2010-08-17 14:31:13 -04:00
Richard Alam
426e9ebac1 - fix timestamps for RTMP audio 2010-08-17 10:35:30 -04:00
Richard Alam
3fef44952a - set ptime:120 and framesPerPacket=6 2010-08-16 16:50:35 -04:00
Richard Alam
b03ffcf30e Merge branch 'master' of github.com:bigbluebutton/bigbluebutton 2010-08-16 15:43:24 -04:00
Richard Alam
862a88c712 - try ptime:40 2010-08-16 15:42:10 -04:00
Sebastian
50332e0e12 Removed the line sip.server.host=ip-here because it was a double entry 2010-08-16 14:39:14 -04:00
Richard Alam
51f0e3237d make it a debug log 2010-08-11 05:12:12 -04:00
Richard Alam
661b0c8f3a - add some println to investigate why speex audio is choppy on Amazon EC2 2010-08-11 04:41:52 -04:00
Richard Alam
1c10da4b8e - put audio packet receive and transcoding into its own thread 2010-08-10 07:52:36 -04:00
Richard Alam
a92059c08c - remove system println 2010-08-04 10:30:27 -04:00
Richard Alam
0dafe99f84 - cleanup 2010-08-04 09:29:03 -04:00
Richard Alam
fcdf6dfb78 - now works with ulaw 8khz and speex 16khz 2010-08-04 09:25:43 -04:00
Richard Alam
6f9bc895ec - dynamically choose between SPEEX and PCMU codec. PCMU codec audio is still choppy. 2010-08-04 07:29:21 -04:00
Richard Alam
b2a56e8926 - cleanup 2010-08-04 06:47:17 -04:00
Richard Alam
ee55647d4b - speex works with refactored transcoders 2010-08-04 06:43:12 -04:00
Richard Alam
3f16b01126 - works great with echo app but not with conference. 2010-07-30 15:41:20 -04:00
Richard Alam
564f22c11b - listen audio stream is good...the talks stream is bad 2010-07-28 17:58:04 -04:00
Richard Alam
0326506df6 - can now make calls using speex. need to improve audio from fp to sip 2010-07-28 16:04:35 -04:00
Richard Alam
c847a4ab3b - connect to local freeswitch 2010-07-23 11:39:28 -04:00
Richard Alam
f305c555f8 - fix formatting 2010-07-22 14:15:52 -04:00
Richard Alam
520d1a3cfc - change copyToLib to resolveDeps 2010-07-21 16:25:47 -04:00
Leif Jackson
8528b2d287 Merge commit 'bbb/master' 2010-07-14 02:53:29 +00:00
Richard Alam
ec256e7cb7 - upgrading versions from 0.64 to 0.7 2010-07-13 09:42:13 -04:00
BigBlueButton
4c1442589d - putting in fix from Leif for bug where session descriptor must not have spaces. 2010-07-11 20:01:06 +00:00
Leif Jackson
f236dc5c1c Merge commit 'bbb/master' sync with bbb 2010-07-11 04:28:39 +00:00
Richard Alam
eee35a0920 - add license header 2010-07-09 15:53:58 -04:00
Richard Alam
7ff9001ba8 - remove error log as it's not an error to get the exception. The exception just signifies 
the call has hangup
 2010-07-06 16:35:34 -04:00
Leif Jackson
ac764dca99 Bug with SIP calls from bbb-voice, Session descriptor cannot have spaces.. Username is not forced to be compliant with SIP Spec 
RFC 2327 username field of owner in sdp cannot contain spaces. Users can login with Spaces in username.
 2010-07-01 07:31:57 +00:00