- a771a51 Ticket #110: HUGE HUGE changeset to replace all occurence of year 2006 with 2007 in copyright notice in all sources by Benny Prijono · 18 years ago
- fc24e69 Fixed various gcc warnings especially related to ticket #82 when particular codecs are disabled by Benny Prijono · 18 years ago
- e960bb5 Fixed gcc warnings, especially about dereferencing type-punned pointer by Benny Prijono · 18 years ago
- 5b1e14d Fixed bug in siprtp: media transport was incorrectly NULL-ed after end of call, causing next call to fail by Benny Prijono · 18 years ago
- af1bb1e Fixed handles leak upon program exit, by introducing pj_shutdown() and pj_atexit(). Also fixed handle leaks in SIP transaction layer and SIP endpoint. by Benny Prijono · 18 years ago
- 5a9e204 Fixed transport detachment when there's no thread being used in siprtp. by Benny Prijono · 18 years ago
- d95abb4 Fixed bug in siprtp in SDP generation because of recent change in SDP attribute value by Benny Prijono · 18 years ago
- 41c96f3 Changed siprtp to bind to specific address (instead of 0.0.0.0) when -i option is specified in the command line by Benny Prijono · 18 years ago
- f521eb0 Fixed assertion error if ACK is received before INVITE transaction sends final response (malicious?). Also fixed misc warnings, and stress-tested on Quad Xeon by Benny Prijono · 18 years ago
- 7bef0f5 Changed siprtp on Linux to raise the thread priority by Benny Prijono · 18 years ago
- 05784a5 Fixed minor bug in siprtp: in summary report, percentage of RX loss is displayed in the TX section by Benny Prijono · 18 years ago
- 258ece9 Few changes in siprtp: by Benny Prijono · 18 years ago
- 513795f Undo modification to use timer, and back to using threads as the performance is not good by Benny Prijono · 18 years ago
- ac623b3 Change all pj_memset to pj_bzero(), where applicable by Benny Prijono · 18 years ago
- ca4cff2 Added INVITE request verification in siprtp before incoming call is established by Benny Prijono · 18 years ago
- b12106f Improvements in PJMEDIA to support RFC 3605 (RTCP attribute in SDP) and other changes to improve RTCP communication behind NAT. Also fixed bug related to RTCP reporting changes in revision 565 by Benny Prijono · 18 years ago
- e695699 Changed siprtp to strictly use one worker thread for all (SIP and media) by Benny Prijono · 18 years ago
- 4d3aa92 Changed siprtp to use timer to schedule transmissions of RTP/RTCP packets by Benny Prijono · 18 years ago
- deb3196 Changed siprtp to use media transport framework to handle NAT by Benny Prijono · 18 years ago
- 1ec70b3 Yet again large diffs because of documentation/doxygen update by Benny Prijono · 18 years ago
- 97f2a37 Changed codec names to uppercase in siprtp, as they are not accepted by some UA by Benny Prijono · 18 years ago
- 6647d82 Set media thread in siprtp samples to highest priority to make RTCP jitter calculation as accurate as possible by Benny Prijono · 18 years ago
- fcb3672 Added --report-file options in siprtp by Benny Prijono · 18 years ago
- b04c9e0 Major modification in pjmedia to split stream transport into separate functionality, to allow using custom transports with streams by Benny Prijono · 18 years ago
- 4d7fd20 Fixed bug in siprtp because of recent changes in pjsip_session_info_from_sdp() by Benny Prijono · 18 years ago
- 8befd9f Another major modifications in PJMEDIA: by Benny Prijono · 18 years ago
- 16a6b0e Split reporting capability to separate file in siprtp sample by Benny Prijono · 18 years ago
- 410fbae Added maximum duration and auto-quit option in siprtp sample. by Benny Prijono · 18 years ago
- 1595301 Initial support for stereo codecs, and added L16 codecs. Also better handling for case remote media is restarted by Benny Prijono · 18 years ago
- 6d7a45f Better support for continuing media when peer has restarted transmission/RTP session by Benny Prijono · 18 years ago
- bf13fee Added stream.c example to demonstrate how to stream audio from both sound device or WAV file by Benny Prijono · 18 years ago
- 393e5f3 Fixed small typo error in printed message in siprtp.c by Benny Prijono · 18 years ago
- 14b7b66 Randomize RTCP send interval to reduce collision risk by Benny Prijono · 18 years ago
- e7483e5 Fixed RTCP compilation warnings with gcc by Benny Prijono · 18 years ago
- 28903eb Beautify screen output of the siprtp sample by Benny Prijono · 18 years ago
- 5b3b460 Fixed crash bug in siprtp sample when call is not connected by Benny Prijono · 18 years ago
- 6996823 Integrate (stream) quality monitoring into RTCP framework, and update all RTCP clients accordingly by Benny Prijono · 18 years ago
- c323807 Protect RTCP packet lost calculation against incorrect packet expectations by Benny Prijono · 18 years ago
- d7a13f1 Added detection/protection against inaccurate timestamp in RTCP by Benny Prijono · 18 years ago
- 49ce9a7 Fixed bug in RTT calculation in RTCP and increase RTT resolution to usec by Benny Prijono · 18 years ago
- a1d03b4 Added RTCP end to end delay calculation by Benny Prijono · 18 years ago
- 46ca996 Fixed bug in RTCP: packing error causing size mismatch when compiled with MSVC by Benny Prijono · 18 years ago
- 9a0eab5 Changed RTCP timing to use high resolution timestamp by Benny Prijono · 18 years ago
- 9d8a873 Fixed SSRC generation in siprtp sample by Benny Prijono · 18 years ago
- 4adcb91 Added more stats and options in siprtp samples by Benny Prijono · 18 years ago
- 60b980e Added siprtp sample by Benny Prijono · 18 years ago