Richard Alam
892de246d3
- cleanup
2012-03-05 15:25:02 +00:00
Richard Alam
edc494ff97
connect to 127.0.0.1 to freeswitch
2012-03-03 10:35:50 -08:00
Richard Alam
1437374a0c
- trim paramaters for sip server
2012-01-25 21:10:50 +00:00
Richard Alam
427f827ba6
- allow users with no mic to listen to audio stream
2012-01-24 21:53:52 +00:00
Richard Alam
6bf086f27b
- fail fast when an attempt to join the voice conference is made but we failed to register with FS.
2012-01-09 21:51:54 +00:00
Richard Alam
e7c1d5e334
- call FS anyway even if red5 hasn't registered
2012-01-09 16:42:43 +00:00
Richard Alam
5ea0e41b2d
upgrading to fixed red.jar for rtmpt
2011-11-16 21:28:09 +00:00
Richard Alam
bb86f44150
- compile with red5-r4293
...
- still some errors to resolve
- had to add aop jars in bbb-apps
- deskshare is choking on tunneling
- exceptions when starting red5 manually
2011-11-04 02:07:35 +00:00
Richard Alam
b795d159f7
build bbb-voice with red5 r4293
2011-11-03 20:24:50 +00:00
Markos Calderon
0e26498b49
changed bbb-voice to use the latest version of red5
2011-10-28 13:03:41 -07:00
Richard Alam
3fa22c6909
- changing build dependencies
2011-10-19 10:34:15 -04:00
Fred Dixon
09d72242f8
Starting to add logic in bbb-conf to help debug record and playback
2011-06-26 10:42:31 -07:00
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
Leif Jackson
9decad03c6
Inital import of freeswitch intergration
2010-06-29 04:51:31 +00:00
Richard Alam
501717666f
- remove extra sip users
...
- rename a few properties
2010-06-24 10:06:55 -04:00
Richard Alam
9fb41dc1e4
- retry 3 times to get an audio port before failing
2010-06-23 16:51:45 -04:00
Richard Alam
ab9caa1616
- add webVoiceConf API so third parties can pass in different extension fo voip
2010-06-22 11:24:22 -04:00
Richard Alam
ea07dba543
- putting timestamp into audio...if connection to a client is slow...audio packets are dropped.
...
(Which one do we prefer? Dropped audio or lag?)
2010-06-21 16:44:38 -04:00
Richard Alam
b8c1443708
- a bit of cleanup
2010-06-21 16:30:35 -04:00
Richard Alam
30251f0700
- has end-to-end call...just need to close rtp ports properly
2010-06-21 15:13:00 -04:00
Richard Alam
e4d905ba49
- can now call into echo test app
2010-06-21 13:08:41 -04:00
BigBlueButton
7128c539c9
- cleaning up a bit
2010-06-18 23:08:20 +00:00
BigBlueButton
7698d77701
- rename stream classes to make it more clear
2010-06-18 21:45:09 +00:00
BigBlueButton
e798f4ad52
- add build info and test tracking branch again
2010-06-18 21:00:29 +00:00
BigBlueButton
d825e3e0b2
- test tracking branch
2010-06-18 20:42:52 +00:00
Richard Alam
e8795dfc63
- continue refactoring
2010-06-18 16:19:19 -04:00
Richard Alam
cd5bc8c8d7
- move things around
2010-06-18 11:46:03 -04:00
Richard Alam
be908218e9
- checkpoint 4: just moving things around
2010-06-18 11:08:31 -04:00
Richard Alam
964577c62e
- registers OK
2010-06-17 15:55:58 -04:00
Richard Alam
7bd89aaab8
- checkpoint 2: continuing refactoring of voice app
2010-06-17 14:16:29 -04:00
Richard Alam
3940207ee4
- checkpoint: refactoring voip app
2010-06-17 11:14:55 -04:00
Richard Alam
5d005b858f
- removing IDE specific files
2010-06-15 16:07:48 -04:00
Richard Alam
9f897c7534
- adding build and lib folder while ignoring their contents. This way, the build and lib folder
...
won't appear in other branches which happens if we just .gitignore them
2010-06-15 15:56:20 -04:00
Richard Alam
b880bbca36
- add files and directories to .gitignore
...
- fix problem where webcam icon stays disabled when webcam is closed
2010-05-31 10:36:15 -04:00
Richard Alam
2e4c441ad7
remove system println
...
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@4324 af16638f-c34d-0410-8cfa-b39d5352b314
2010-05-14 19:10:17 +00:00
Richard Alam
6e84e39387
- upgrade to red5-0.91
...
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@4051 af16638f-c34d-0410-8cfa-b39d5352b314
2010-03-29 18:19:22 +00:00
Richard Alam
fca902fd21
- change default ports to unassigned ports defined by IANA
...
http://www.iana.org/assignments/port-numbers
- increase default users to 100
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3998 af16638f-c34d-0410-8cfa-b39d5352b314
2010-03-24 19:41:25 +00:00
Richard Alam
ea6addab2f
- upgrade to Red5 0.9
...
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3741 af16638f-c34d-0410-8cfa-b39d5352b314
2010-02-19 17:59:12 +00:00
Richard Alam
8a1250e46b
- fix how rtpport and sipport are incremented
...
- change sipport from 5070-5099 to 6070-6099. We get BindException on 5080 because 5080 is already used by Red5.
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3101 af16638f-c34d-0410-8cfa-b39d5352b314
2009-12-15 02:43:55 +00:00
Richard Alam
e69587f50d
- close socket properly
...
- modify threading model
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3100 af16638f-c34d-0410-8cfa-b39d5352b314
2009-12-14 23:26:47 +00:00
Richard Alam
d50467ecd6
- rename things
...
- don't close socket too early when quitting as we get IOException when trying to receive RTP packets
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3094 af16638f-c34d-0410-8cfa-b39d5352b314
2009-12-14 18:52:30 +00:00
Richard Alam
466f1a11e1
- add comment to not put timestamp on outgoing audio packet to flash player
...
as it results in choppy audio
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3087 af16638f-c34d-0410-8cfa-b39d5352b314
2009-12-10 21:36:07 +00:00
Richard Alam
be9d237bd6
- minimize logging
...
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3082 af16638f-c34d-0410-8cfa-b39d5352b314
2009-12-08 17:28:17 +00:00
Richard Alam
20ddea06eb
- change logging appender to rolling file
...
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3078 af16638f-c34d-0410-8cfa-b39d5352b314
2009-12-07 20:13:05 +00:00
Richard Alam
6dd9c14fd6
- put sip log logging level into bigbluebutton-sip.properties
...
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3076 af16638f-c34d-0410-8cfa-b39d5352b314
2009-12-07 20:08:53 +00:00
Richard Alam
d0aad9bd17
- modify how the listen stream gets started/stopped which may be the cause of choppy audio
...
- make Sip stack logging configurable from bigbluebutton.properties
- cleanup
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3074 af16638f-c34d-0410-8cfa-b39d5352b314
2009-12-07 18:57:30 +00:00
Richard Alam
4391bab0c3
- inject SipManager using Spring
...
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3070 af16638f-c34d-0410-8cfa-b39d5352b314
2009-12-04 23:29:10 +00:00
Richard Alam
6789a975c8
- cleanup and move things around
...
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3045 af16638f-c34d-0410-8cfa-b39d5352b314
2009-11-28 14:11:56 +00:00
Richard Alam
700e919305
- distinguish between REGISTER and UNREGISTER
...
- this will fix where the client disconnects/connects when the UA renews registration
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3042 af16638f-c34d-0410-8cfa-b39d5352b314
2009-11-27 18:30:27 +00:00
Richard Alam
1afb323411
- display the username instead of the userid on the voice participant's window
...
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3040 af16638f-c34d-0410-8cfa-b39d5352b314
2009-11-25 21:04:02 +00:00
Richard Alam
a947093768
- oops...forgot to remove hardcoded number for echo test
...
git-svn-id: http://bigbluebutton.googlecode.com/svn/trunk@3039 af16638f-c34d-0410-8cfa-b39d5352b314
2009-11-25 19:10:09 +00:00