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191 Commits

Author SHA1 Message Date
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