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Timeline
Feb 5, 2008:
- 9:17 PM ARM_QEMU edited by
- (diff)
- 8:40 PM ARM_QEMU edited by
- (diff)
- 7:52 PM Ticket #468 (Support for non-looping playback in memory player) created by
- 6:25 PM Ticket #467 (RTP/AVP vs RTP/SAVP negotiation) created by
- Issues: 1. Currently, when pjsip sends offer with RTP/AVP with a=crypto …
- 2:49 PM ARM_QEMU edited by
- (diff)
- 1:54 AM ARM_QEMU edited by
- (diff)
- 12:00 AM ARM_QEMU edited by
- (diff)
Feb 4, 2008:
- 11:56 PM Changeset [1779] by
- Ticket #461: Initial integration of the new jitter buffer into stream
- 11:52 PM Changeset [1778] by
- Ticket #461: - added param userdata to JB callbacks - added a bit …
- 11:44 PM ARM_QEMU edited by
- (diff)
- 11:37 PM ARM_QEMU edited by
- (diff)
- 11:37 PM Ticket #466 (Speex decoder set decoded frame size incorrectly) closed by
- fixed: In r1777: Fixed PCM frame size returned by speex decoder.
- 11:37 PM Changeset [1777] by
- Ticket #466: Fixed PCM frame size returned by speex decoder.
- 11:34 PM Ticket #466 (Speex decoder set decoded frame size incorrectly) created by
- It was hardcoded to 320 bytes (160 samples) regardless what mode the speex …
- 3:11 PM ARM_QEMU created by
- 7:19 AM FAQ edited by
- (diff)
- 7:09 AM FAQ edited by
- (diff)
Feb 3, 2008:
- 4:59 PM Changeset [1776] by
- Ticket #461: - added feature resample, to test the effect of drift …
- 4:55 PM Changeset [1775] by
- Ticket #461: - added JB phases(idle, learning, running) - changed …
Feb 2, 2008:
- 5:07 PM Changeset [1774] by
- More ticket #61: fixed signaling security level calculation for SRTP
- 4:37 PM Changeset [1773] by
- Ticket #461: Fixed bug in find_matched_window(), the ref pointer is not …
- 12:12 PM Changeset [1772] by
- Ticket #461: Added another test for JB2, which the objective is to measure …
- 11:49 AM Changeset [1771] by
- Ticket #461: Added test for new jitter buffer (JB2). The test will read …
- 11:23 AM Changeset [1770] by
- Ticket #461: Added the new jitter buffer, initial source.
- 9:12 AM Changeset [1769] by
- More ticket #465: option to return UDP header in pj_pcap_read_udp(), and …
- 9:09 AM Changeset [1768] by
- Fixed warning about uninitialized tpinfo variable
- 8:22 AM Changeset [1767] by
- Added trace to PCAP reader
Feb 1, 2008:
- 4:44 PM Changeset [1766] by
- More ticket #465: updated VS8 and WinCE projects
- 2:59 PM Ticket #465 (New utility to read RTP packets/payload from PCAP file) closed by
- fixed: Done in r1765
- 2:59 PM Changeset [1765] by
- Ticket #465: utility to read PCAP file and sample application to read RTP …
- 2:57 PM Ticket #465 (New utility to read RTP packets/payload from PCAP file) created by
- Included in this package: - new pcap.[hc] file in PJLIB-UTIL to …
- 2:53 PM Changeset [1764] by
- More ticket #61: mismatch in libsrtp output library name causing error …
- 2:51 PM SRTP edited by
- (diff)
- 2:47 PM Changeset [1763] by
- More ticket #61: removed const from rtp and rtcp callback function …
- 9:34 AM FAQ edited by
- (diff)
Jan 31, 2008:
- 9:26 PM Changeset [1762] by
- Added media transport diagram for documentation
- 9:59 AM FAQ edited by
- (diff)
- 8:58 AM FAQ edited by
- (diff)
- 8:53 AM FAQ edited by
- (diff)
- 8:11 AM Ticket #464 (Jitter buffer should return length information) closed by
- fixed: Committed in r1761
- 8:11 AM Changeset [1761] by
- Ticket #464: jitter buffer should return frame length information
Jan 30, 2008:
- 5:03 PM Ticket #464 (Jitter buffer should return length information) created by
- Currently jitter buffer doesn't return length information, which causes …
- 2:46 PM Ticket #463 (Configuration option to have external sound device implementation) created by
- Currently pjmedia allows selection of audio backend via build settings …
- 2:45 PM Ticket #462 (Playback of PCAP file) created by
- 9:47 AM Changeset [1760] by
- Ticket #61: undo r1759 and replace it with other patches to fix unable to …
Jan 29, 2008:
- 6:35 PM Ticket #461 (New Jitter Buffer Development) created by
- Main objectives are aiming optimum latency, drift handling, less noise, …
- 5:40 PM Changeset [1759] by
- Ticket #61: bug, unable to accept RTP/AVP with crypto attribute when …
- 11:52 AM Changeset [1758] by
- Ticket #61: --srtp-use option in pjsua did not change account config
Jan 27, 2008:
- 10:28 PM Ticket #460 (Concurrency problem when destroying stream (thanks Michael Broughton)) created by
- [http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2008-January/001463 …
Jan 26, 2008:
- 10:45 AM Changeset [1757] by
- Added pj_strstr() and pj_stristr() in pjlib
- 10:45 AM Changeset [1756] by
- Bug: source address not initialized in loop transport causing pjsip-test …
- 10:18 AM Changeset [1755] by
- Move branches/pjproject-0.5-stable to archives
- 10:17 AM Changeset [1754] by
- Move branches/trunk-0.6-pre to archives
- 10:16 AM Changeset [1753] by
- Move branches/pjproject-0.2 to archives
- 10:15 AM Changeset [1752] by
- Added branches/archives directory
- 10:09 AM Changeset [1751] by
- Branched trunk to branches/projects/jbuf2 for new jitter buffer …
- 10:07 AM Changeset [1750] by
- Branched trunk to branches/projects/turn, for TURN-06 and ICE and new …
- 10:05 AM Changeset [1749] by
- Added branches/projects directory
Jan 25, 2008:
- 10:01 PM SRTP edited by
- (diff)
- 4:06 PM Changeset [1748] by
- Fixed doxygen comments everywhere
- 2:51 PM SRTP edited by
- (diff)
- 2:49 PM SRTP edited by
- (diff)
- 12:01 PM Ticket #459 (pjmedia_clock_create() should not create thread when ...) closed by
- fixed: Patch committed in r1747. Thanks.
- 12:01 PM Changeset [1747] by
- Ticket #459: pjmedia_clock_create() should not create thread when …
- 11:47 AM Ticket #459 (pjmedia_clock_create() should not create thread when ...) created by
- Creating pjmedia_clock always create background thread, regardless the …
- 11:28 AM Ticket #61 (Support for SRTP) closed by
- fixed: r1746: - disable CPU_RISC as it seems to be broken (tested on ARM)
- 11:28 AM Changeset [1746] by
- More ticket #61: disable CPU_RISC in SRTP as it seems to be broken (tested …
- 9:41 AM SRTP edited by
- (diff)
- 9:39 AM SRTP edited by
- (diff)
- 9:02 AM Changeset [1745] by
- More ticket #61: fix bug in secure signaling determination, and added …
Jan 24, 2008:
- 8:50 PM SRTP edited by
- (diff)
- 8:36 PM FAQ edited by
- (diff)
- 8:01 PM Ticket #458 (PJ_ERESOLVE error in Windows Mobile) closed by
- fixed
- 7:59 PM Changeset [1744] by
- Related to ticket #458: pj_getaddrinfo() should return success if IP …
- 7:54 PM Changeset [1743] by
- Related to ticket #458: enum_interface() should return non-PJ_SUCCESS if …
- 7:20 PM Changeset [1742] by
- Return 406/Not Acceptable if SRTP negotiation failed instead of 500
- 7:20 PM Changeset [1741] by
- Fixed libsrtp random string generation on Win32
- 6:35 PM Changeset [1740] by
- More ticket #61: error detecting CPU type in srtp_config.h causing libsrtp …
- 3:29 PM Changeset [1739] by
- More ticket #61: fix potential error when looking up SRTP error string, …
- 3:27 PM Changeset [1738] by
- More ticket #61: SRTP will try to use /dev/urandom as RNG if fcntl.h and …
- 3:18 PM Changeset [1737] by
- More ticket #61: crashed on SRTP error string lookup
- 3:14 PM Ticket #458 (PJ_ERESOLVE error in Windows Mobile) created by
- See mailing list discussions: - …
- 10:23 AM Changeset [1736] by
- Ticket #61: added Windows Mobile/WinCE eVC project for SRTP and libsrtp
- 9:04 AM FAQ edited by
- Added SRTP entry and fixed broken links (diff)
- 8:59 AM TLS edited by
- (diff)
- 8:23 AM WikiStart edited by
- (diff)
- 7:35 AM SRTP edited by
- (diff)
Jan 23, 2008:
- 8:39 PM Changeset [1735] by
- Ticket #61: Implement SRTP support in PJMEDIA and PJSUA-LIB, and updated …
- 8:34 PM Changeset [1734] by
- Handle SDP offer containing application as the media type
- 8:29 PM Changeset [1733] by
- Related to ticket #61: added new invite session API …
- 8:25 PM Changeset [1732] by
- Related to ticket #61: Added PJSIP_ESESSIONINSECURE error code in …
- 8:23 PM Changeset [1731] by
- Fixed integer conversion/format warning on x86_64 Linux
- 8:17 PM Changeset [1730] by
- Ticket #61: imported libsrtp into third_party directory (backported from …
- 7:58 PM SRTP edited by
- (diff)
- 6:49 PM SRTP edited by
- (diff)
- 6:01 PM SRTP edited by
- (diff)
- 4:20 PM SRTP edited by
- (diff)
- 4:03 PM SRTP created by
- 2:36 PM Ticket #452 (SRTP development) closed by
- fixed: Moved to TRUNK!
- 2:34 PM Changeset [1729] by
- Merged from SRTP branch: added stdint.h header detection in autoconf. …
- 1:42 PM Changeset [1728] by
- Added stdint.h detection (PJ_HAS_STDINT_H) and minor fix in …
- 12:16 PM FAQ edited by
- (diff)
- 11:03 AM Changeset [1727] by
- Fixed various warnings when building on x86_64 Linux target
- 8:23 AM FAQ edited by
- (diff)
- 8:16 AM FAQ edited by
- (diff)
Jan 22, 2008:
- 10:31 PM Changeset [1726] by
- Ticket #452: - added part to test specific key, only for testing purpose - …
- 5:08 PM SIP_Redirection_Testing edited by
- (diff)
- 5:05 PM FAQ edited by
- (diff)
- 5:00 PM TLS edited by
- (diff)
- 4:52 PM TLS edited by
- (diff)
- 4:29 PM TLS edited by
- (diff)
- 3:07 PM TLS edited by
- (diff)
- 3:07 PM TLS edited by
- (diff)
- 3:06 PM TLS created by
- 12:33 PM PJSIP_IMS edited by
- (diff)
- 12:31 PM PJSIP_IMS edited by
- (diff)
- 12:22 PM PJSIP_IMS created by
- 11:59 AM Ticket #457 (Add new media type: application) closed by
- fixed: r1725: Added media type 'application'.
- 11:58 AM Changeset [1725] by
- Ticket #457: Added media type 'application'.
- 11:56 AM Ticket #457 (Add new media type: application) created by
- Avoiding incoming call rejected because of unrecognized media type.
- 11:04 AM SIP_Redirection_Testing created by
- 10:55 AM SIP_Redirection edited by
- (diff)
- 10:48 AM SIP_Redirection edited by
- (diff)
- 10:43 AM SIP_Redirection edited by
- (diff)
- 9:46 AM SIP_Redirection created by
- 1:56 AM Changeset [1724] by
- Ticket #452: Bug fixed in comparing crypto name in media_start.
Jan 21, 2008:
- 9:20 PM Changeset [1723] by
- Ticket #452: - Applied preprocessor PJMEDIA_HAS_SRTP in PJMEDIA & PJSUA. - …
- 6:08 PM Changeset [1722] by
- Ticket #452: - Bug fixed in mandatory mode, change local media sdp …
- 3:28 AM Changeset [1721] by
- Ticket #452: bugs fixed in transport_srtp.c: - default crypto-suites in …
- 2:25 AM Changeset [1720] by
- Ticket #452: Added fields use_srtp and srtp_secure_signaling to …
Jan 20, 2008:
- 10:35 PM Ticket #456 (Option to follow redirection in PJSUA-LIB) created by
- Implement handling of call redirection in PJSUA-LIB: - a callback to …
- 2:28 PM Changeset [1719] by
- Ticket #452: renamed SRTP config.h to srtp_config.h, added srtp_err.c to …
Jan 19, 2008:
- 3:51 PM Changeset [1718] by
- More work on ticket #452/SRTP. Please see this ticket for detailed …
- 1:02 PM Ticket #455 (Allocate call id in round robin fashion) closed by
- fixed: Done in r1717
- 1:01 PM Changeset [1717] by
- Ticket #455: allocate pjsua call id in round robin fashion
- 12:56 PM Ticket #455 (Allocate call id in round robin fashion) created by
- Currently call id allocation is done by searching for empty slot from …
- 12:22 PM Changeset [1716] by
- Added information about delaybuf result on MacOS X 10.5
- 9:39 AM Changeset [1715] by
- More work on ticket #438: Workaround for frame bursts from audio devices. …
- 9:34 AM Changeset [1714] by
- Fixed warning about uninitialized variable in VS6
- 1:42 AM Changeset [1713] by
- Ticket #452: changed on transport_srtp.c: (re)applied setting close …
- 1:36 AM Changeset [1712] by
- Ticket #452: changed libsrtp error string resolver
Jan 18, 2008:
- 8:42 PM Changeset [1711] by
- - implemented new flow based on new SRTP transport interface and data …
- 8:14 PM Changeset [1710] by
- Ticket #452: Updated errno.h & .c to accomodate SRTP Transport & libsrtp …
- 7:24 PM Ticket #454 (WMME assertion after running application for a while) created by
- One user reported that after running application with WMME backend for a …
- 6:49 PM Changeset [1709] by
- More ticket #452: SRTP creation/destruction, added use_srtp and …
- 6:38 PM Changeset [1708] by
- Ticket #452: updated ICE media transport's media_create() to consider the …
- 6:36 PM Changeset [1707] by
- Added PJSIP_ESESSIONINSECURE error code in sip_errno to require that …
- 5:18 PM Changeset [1706] by
- Fixed python module build error (thanks Jeff Ames)
- 1:25 PM Ticket #453 (Log level is not set in PJSUA-LIB (thanks Simon Farmer)) closed by
- fixed: Fixed in r1705
- 1:25 PM Changeset [1705] by
- Ticket #453: Log level is not set in PJSUA-LIB (thanks Simon Farmer)
- 1:24 PM Changeset [1704] by
- Minor correction about WAV player parameter name and its comment in …
- 1:24 PM Changeset [1703] by
- Minor correction about WAV player parameter name and its comment in …
- 1:20 PM Ticket #453 (Log level is not set in PJSUA-LIB (thanks Simon Farmer)) created by
- The log level setting in pjsua_logging_config in PJSUA-LIB is not …
- 12:30 PM Changeset [1702] by
- Ticket #452: changed SRTP API with new specification from test doc
Jan 17, 2008:
- 8:48 PM Changeset [1701] by
- put a note about disabling the 'rand_s' block in libsrtp file …
- 8:37 PM Changeset [1700] by
- Ticket #452: - change libsrtp on rand_source.c to disable rand_s which …
- 7:17 PM Changeset [1699] by
- Continuing ticket #452: added VS6 project for libsrtp, and changed libsrtp …
- 5:29 PM Changeset [1698] by
- Ticket #452: - Add directory srtp to third_party/build directory - Add …
- 4:20 PM Changeset [1697] by
- Ticket #452: imported libsrtp library into third_party directory
- 3:08 PM Ticket #452 (SRTP development) created by
- 2:09 PM Changeset [1696] by
- Branched trunk to branches/users/nanang
- 2:07 PM Changeset [1695] by
- Added branches/users directory
- 10:14 AM FAQ edited by
- (diff)
- 10:05 AM SIP_Message_Buffer_Event edited by
- (diff)
- 10:02 AM SIP_Message_Buffer_Event edited by
- (diff)
- 9:57 AM SIP_Message_Buffer_Event edited by
- (diff)
- 9:08 AM SIP_Message_Buffer_Event edited by
- (diff)
- 8:53 AM SIP_Message_Buffer_Event created by
- 6:21 AM PJSIP_Tutorial edited by
- (diff)
Jan 16, 2008:
- 9:13 AM Ticket #451 (Updated PortAudio to the latest revision) created by
- This fixes several problems, among other things: 1. Minor bug in …
- 9:07 AM Ticket #450 (PJLIB function to query the DNS server configured in the system) created by
- Currently PJLIB lacks a portable function to query what is the DNS server …
Jan 15, 2008:
- 8:26 PM Changeset [1694] by
- Fixed compilation warnings on OSX 10.5
- 4:54 PM Changeset [1693] by
- Reply with 488+SDP instead of 415 when incoming SDP is not acceptable …
Jan 14, 2008:
- 6:35 PM Changeset [1692] by
- Related to ticket #449: fixed bug when rx volume adjustment caused static …
- 4:18 PM Ticket #449 (New conference mixing algorithm) closed by
- fixed: Done in r1691
- 4:18 PM Changeset [1691] by
- Ticket #449: New conference mixing algorithm to replace multiplication …
- 4:17 PM Ticket #449 (New conference mixing algorithm) created by
- New mixing algorithm for conference bridge: rather than doing …
- 4:12 PM Changeset [1690] by
- Related to ticket #447: the WAV player should consistently return …
- 1:30 PM Ticket #448 (PJSIP rejects incoming call with 415/Unsupported Media Type for INVITE ...) closed by
- fixed: Fixed in r1689
- 1:30 PM Changeset [1689] by
- Ticket #448: PJSIP rejects incoming call with 415/Unsupported Media Type …
- 1:29 PM Ticket #448 (PJSIP rejects incoming call with 415/Unsupported Media Type for INVITE ...) created by
- When INVITE contains video line, PJSIP will reject it with 415, while it …
- 11:54 AM Changeset [1688] by
- Fixed Symbian include header case sensitivity (thanks Alfred E. Heggestad)
Jan 13, 2008:
- 12:26 AM Changeset [1687] by
- Renamed symbian_ua.pkg to symbian_ua_udeb.pkg and symbian_ua_urel.pkg
- 12:25 AM DevelopingSymbianAppWithCarbide edited by
- (diff)
Jan 12, 2008:
- 3:32 PM Changeset [1686] by
- In mix.c, handle FRAME_TYPE_NONE if the conference ever returns one
- 3:30 PM Changeset [1685] by
- Another fix for ticket #447: WAV player cuts last data from the WAV file
- 2:16 PM Changeset [1684] by
- Fixed overflow bug in length calculation, added force option, and added …
- 2:14 PM Ticket #447 (WAV player cuts last data from the WAV file) closed by
- fixed: Fixed in r1683
- 2:13 PM Changeset [1683] by
- Ticket #447: WAV player cuts last data from the WAV file
- 2:13 PM Ticket #447 (WAV player cuts last data from the WAV file) created by
- When the WAV player (pjmedia_wav_player_port) is not configured to …
- 11:32 AM Ticket #446 (Added sample application to mix WAV files using conference bridge) closed by
- fixed: Done in r1681 and r1682
- 11:31 AM Changeset [1682] by
- Ticket #446: Added sample application to mix WAV files using conference …
- 11:23 AM Changeset [1681] by
- Ticket #446: Added sample application to mix WAV files using conference …
- 11:22 AM Ticket #446 (Added sample application to mix WAV files using conference bridge) created by
- A new sample file, mix.c (under pjsip-apps/src/samples directory, …
- 11:11 AM Changeset [1680] by
- Added new WAV player API: pjmedia_wav_player_get_len()
Jan 11, 2008:
- 4:30 PM Changeset [1679] by
- Updated encdec sample to support PLC etc
- 9:02 AM Ticket #445 (Added codec encoding/decoding sample) closed by
- fixed: Done in r1678
- 9:01 AM Changeset [1678] by
- Ticket #445: added codec encoding/decoding sample
- 9:00 AM Ticket #445 (Added codec encoding/decoding sample) created by
- A simple example on how to use the codec. The sample will read the PCM …
- 8:52 AM Changeset [1677] by
- Fixed what looks like a bug in pjmedia's Speex decoder: decoder may return …
- 8:24 AM Ticket #444 (Bug in tone generator: can't play more digits (thanks Marian Dragomir)) closed by
- fixed: Fixed in r1676. Also this commit added lock protection (can be disabled …
- 8:22 AM Changeset [1676] by
- Ticket #444: Bug in tone generator: can't play more digits (thanks Marian …
- 8:12 AM Ticket #444 (Bug in tone generator: can't play more digits (thanks Marian Dragomir)) created by
- Bug in tone generator causes it to fail to generate more digits. Thanks …
Jan 10, 2008:
- 12:09 PM Changeset [1675] by
- Added PJSUA_DEFAULT_AUDIO_FRAME_PTIME setting and changed default iLBC …
Jan 9, 2008:
- 10:12 PM Ticket #443 (Overflow in dump_media_session() (thanks Simon Farmer)) closed by
- fixed: Fixed in r1674
- 10:11 PM Changeset [1674] by
- Ticket #443: Overflow in dump_media_session() (thanks Simon Farmer)
- 10:10 PM Ticket #443 (Overflow in dump_media_session() (thanks Simon Farmer)) created by
- The calculations in this function will overflow for longer call period. …
- 4:58 PM Ticket #442 (Function to create dialog/call with specifying Call-ID) created by
- This is useful when creating a new call as the result of redirection, as …
- 3:43 PM Ticket #441 (Fix endianess detection for PortAudio) closed by
- fixed: In r1673: - fix configure script to set PortAudio's endianness
- 3:43 PM Changeset [1673] by
- Ticket #441: Fix endianess detection for PortAudio?
- 3:42 PM Ticket #441 (Fix endianess detection for PortAudio) created by
- We don't use PortAudio's configure script for building PortAudio, and it …
- 3:26 PM Ticket #440 (pj_timer_entry_init() doesn't initialize timer ID) closed by
- fixed: In r1672: - this function now will set _timer_id field of …
- 3:25 PM Changeset [1672] by
- Ticket 440: pj_timer_entry_init() doesn't initialize timer ID
- 3:24 PM Ticket #440 (pj_timer_entry_init() doesn't initialize timer ID) created by
- See the discussion in …
- 10:01 AM FAQ edited by
- (diff)
Jan 8, 2008:
- 11:41 PM Changeset [1671] by
- Protect against division by zero in pjsua's dump_media_session (thanks …
- 10:50 PM Changeset [1670] by
- Added call gaping in siprtp
- 10:17 PM CodecPerformance created by
- 4:03 PM Changeset [1669] by
- Fixed warnings with strict gcc
- 4:02 PM Changeset [1668] by
- Fixed compilation error about IFF_UP
Jan 7, 2008:
- 9:35 PM FAQ edited by
- (diff)
- 9:25 PM FAQ edited by
- (diff)
- 9:20 PM FAQ edited by
- (diff)
- 6:33 PM Ticket #439 (Encoder is called for FRAME_TYPE_NONE) closed by
- fixed: In r1667: - don't call encode() if input frame is FRAME_TYPE_NONE
- 6:33 PM Changeset [1667] by
- Ticket #439: don't call encode() if input is FRAME_TYPE_NONE
- 6:11 PM Ticket #439 (Encoder is called for FRAME_TYPE_NONE) created by
- Ticket #56 makes stream.c calls codec's encode() even when FRAME_TYPE_NONE …
- 1:50 PM Changeset [1666] by
- Assertion error in streamutil sample
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