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2,343 Commits 13 Branches 260 Tags 1.1 GiB
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114 Commits

Author SHA1 Message Date
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