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2,388 Commits 13 Branches 260 Tags 1.1 GiB
2ab97b4727
Commit Graph

127 Commits

Author SHA1 Message Date
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