- 0324ba5 Fixed #1170 (Assertion when receiving updated SDP offer with all media lines removed): by Benny Prijono · 14 years ago
- 3150d8b Fix #1165: by Nanang Izzuddin · 14 years ago
- 1e952a8 Fix #1143: by Nanang Izzuddin · 14 years ago
- 5e39a2b Closed #1129: by Nanang Izzuddin · 14 years ago
- 2dbf507 Fixed #1079 (Media transport should be kept alive during double-hold scenario). Details: by Benny Prijono · 14 years ago
- a8f9e62 Fixes #1047 (Don't send UPDATE if remote doesn't support it (thanks Bogdan Krakowski for the report)) and fixes #1097 (Support sending UPDATE without SDP). Details: by Benny Prijono · 14 years ago
- 4d6ff4d Fixed #1077: In ICE stream transport (ice_strans.c), automaticaly retry allocation once if TURN allocation fails. If this allocation retry also fails, notify the TURN user via on_ice_complete() callback. Details: by Benny Prijono · 14 years ago
- 16852b3 Re #668: by Nanang Izzuddin · 14 years ago
- d89cc3a Re #1069: by Nanang Izzuddin · 14 years ago
- 4e5c3f5 Fixed #1062 (Assertion if 200/OK INVITE response is received during PJSUA-LIB destroy sequence) by Benny Prijono · 14 years ago
- 06839e7 Ticket #1028: Recommit r3074 with updated codec.h. by Nanang Izzuddin · 15 years ago
- 0ff300c Undo r3074 for ticket #1028 as it is missing updated codec.h hence causing build errors (thanks Michael Bradley Jr for the report) by Benny Prijono · 15 years ago
- 7217505 Ticket #1028: by Nanang Izzuddin · 15 years ago
- bf53b00 Misc fix (#1003): resolve NAT type after mapped addresses for SIP/RTP/RTCP sockets have been resolved, so reduce chattiness during initialization and simplify debugging related to STUN problems by Benny Prijono · 15 years ago
- 8ea7eb0 Misc fix (#951): Fixed pjsua bug RTP timestamp & sequence resetted after hold-resume (thanks Nikolay Popok for the report). by Nanang Izzuddin · 15 years ago
- 4d79b0f Initial commit for ticket #950: QoS support: by Benny Prijono · 15 years ago
- f5d9f1f Ticket #881: send UPDATE or re-INVITE after ICE negotiation, if the default candidate has changed by Benny Prijono · 15 years ago
- 384dab4 Ticket #970: More gracefull PJSUA-LIB shutdown sequence. Enhancements: by Benny Prijono · 15 years ago
- 40d62b6 Ticket #877: Memory consumption of the invite session grows indefinitely if call is running for long period of time and with many re-INVITES by Benny Prijono · 15 years ago
- bb995fd Ticket #866: Allow application to specify more than one STUN servers for more robustness, and continue application startup if STUN resolution fails by Benny Prijono · 15 years ago
- e25fe6f Sound device is not automatically started after pjsua_set_no_snd_dev() is called, even after pjsua_set_snd_dev() is called by Benny Prijono · 15 years ago
- 873f3e4 Ticket #919: by Nanang Izzuddin · 15 years ago
- abf58db Ticket #910: by Nanang Izzuddin · 15 years ago
- 7082b26 Ticket #882: Added check of active call count before auto-close sound device. by Nanang Izzuddin · 15 years ago
- d65f78c More ticket #876: - fixed crash when null-audio is used with switchboard by Benny Prijono · 15 years ago
- 23ea21a Ticket #876: Second call fails to open the sound device event when --null-audio is set by Benny Prijono · 15 years ago
- 329d638 Integration of Sipit24 branch, many tickets involved: by Benny Prijono · 15 years ago
- 0f711b4 Ticket #824: Race condition in sound auto-close feature may cause sound device to not be opened by Benny Prijono · 15 years ago
- 69b69ae Ticket #788: Updated pjsua_media_subsys_init() to perform SRTP library initialization. by Nanang Izzuddin · 15 years ago
- 57b8857 Ticket #774: by Nanang Izzuddin · 15 years ago
- e506c8c Disable echo cancellation related setting from pjsua_aud_get/set_setting() by Benny Prijono · 16 years ago
- 09b0ff6 Bug fixes from last changes: by Benny Prijono · 16 years ago
- f798e50 BIG refactoring in pjsua_media.c: by Benny Prijono · 16 years ago
- 8465c68 Fixed pjsua_set_snd_dev() to differentiate the way of opening sound device based on the used conf type. by Nanang Izzuddin · 16 years ago
- 0cb3b02 Added one new pjsua-lib API to get audio device stream instance, also added its usage sample for audio routing in symbian_ua. by Nanang Izzuddin · 16 years ago
- 96e74f3 - New convention about default audio device ID (now there is different ID for default capture/playback device. It should be backward compatible) by Benny Prijono · 16 years ago
- 10454dc Updated libraries and applications to use the new Audio Device API by Benny Prijono · 16 years ago
- 418e0a4 Fixed checking macro in pjsua-lib for Symbian APS audio routing. by Nanang Izzuddin · 16 years ago
- 452b66b - Updated audio route API for Symbian APS. by Nanang Izzuddin · 16 years ago
- fe02a06 - Added APS-direct sound device management into pjsua-lib (and removed it from apps). by Nanang Izzuddin · 16 years ago
- 81db8c7 - Added new API for sound & sound port to create/open sound device with extended setting, to allow opening sound device with non-PCM format and other settings. by Nanang Izzuddin · 16 years ago
- 670f71b Ticket #699: Added sound device idle checking in media_channel_deinit(), which is called each time a call being disconnected (for any reason). by Nanang Izzuddin · 16 years ago
- 844653c Updated copyright notice in all files to Teluu Inc., and changed the year as well by Benny Prijono · 16 years ago
- 6aa4495 Removed 'odd' clock rate from sound device clock rates list. by Nanang Izzuddin · 16 years ago
- d704a8b Fixed bug missing reinit med_orig when reinit pjsua media transports. by Nanang Izzuddin · 16 years ago
- 0378905 Ticket #634: Assertion when rejecting incoming INVITE when the call arrives while pjsip is being shutdown by Benny Prijono · 16 years ago
- 40860c3 Ticket #610: Added sample to create a media transport adapter, similar to how SRTP media transport works by Benny Prijono · 16 years ago
- 437d77c Ticket #602: fixed assertion of invalid stream index supplied to pjmedia_session_get_stream_stat(). by Nanang Izzuddin · 16 years ago
- a815ceb Ticket #602: by Nanang Izzuddin · 16 years ago
- 6df1d53 Ticket #583: Fixed preprocessor check in pjsua_media.c on deinit-ing IPP codec. by Nanang Izzuddin · 16 years ago
- 3c1ae63 Fixed pjsua-lib bug that it failed to connect sound device & conference for some audio_frame_ptime settings (thanks Thomas Ramp). by Nanang Izzuddin · 16 years ago
- a4e7cdd More ticket #583: a bit of tidying up and renamed macro names etc. by Benny Prijono · 16 years ago
- 7dd3268 Ticket #583: Added missing IPP codec deinit and fixed matching #if-#endif in IPP codec encode. by Nanang Izzuddin · 16 years ago
- 493a8db Ticket #583: Initial source of IPP codecs wrapper. by Nanang Izzuddin · 16 years ago
- 551af42 Ticket #586: Added ICE negotiations test to test the scenario when two agents have different number of components by Benny Prijono · 16 years ago
- 99d6952 Ticket #563: Updated SDP offer/answer related to call hold scenario to conform to RFC 3264 section 8.4 (before: 'a=inactive' and 'c=0.0.0.0', now: 'a=sendonly' and muted ports). by Nanang Izzuddin · 16 years ago
- 311b63f Related to ticket #566: Crash when shutting down PJSUA-LIB and outgoing call in in progress and there is no answer from remote by Benny Prijono · 16 years ago
- ab8dba9 Added more Python tests: offer with multiple media lines by Benny Prijono · 16 years ago
- a310bd2 Fix handling of multiple media lines in the incoming SDP offer. Now pjsua-lib will be able to select the best media line to handle by Benny Prijono · 16 years ago
- 81e9bd5 Fixed minor bug on ptime calculation on player creation; Updated pjsua-app info on ports ptime and default capture latency by Nanang Izzuddin · 16 years ago
- 4375f90 Ticket #544: Fixed SRTP on hold+reinvite scenario by Nanang Izzuddin · 16 years ago
- 88accae Specifying star in codec selection will select all codecs. Fixed the codec selection in codec.c to select all codecs when empty string is given. by Benny Prijono · 16 years ago
- 32177c0 Large changeset to replace all occurences of year 2007 with 2008 in the copyright notice by Benny Prijono · 16 years ago
- 224b4e2 Ticket #549: major modification in media transport API to support more offer/answer scenarios by Benny Prijono · 16 years ago
- 148fd39 More ticket #540: updated snd-auto-close to work friendly with call by Nanang Izzuddin · 16 years ago
- 68559c3 Ticket #540: Added pjsua-lib feature auto-close sound device on idle and new pjsua option --snd-auto-close=N by Nanang Izzuddin · 16 years ago
- 9dbad15 Updated default speex quality settings and reenabled pjsua to set Speex codec quality based on media quality config by Nanang Izzuddin · 16 years ago
- fd461eb Added more validations & a new API (thanks Florian Bomers): by Nanang Izzuddin · 16 years ago
- f76e139 More ticket #485: added TURN support in PJSUA-LIB API by Benny Prijono · 16 years ago
- e85a183 Fixed bug wrong option for resample port between conference bridge and sound device by Nanang Izzuddin · 16 years ago
- 829ac02 Changed build optimizations settings for Speex, pjmedia, and symbian_sound for Symbian. Speex/8000 now also runs on Symbian! by Nanang Izzuddin · 16 years ago
- 53a7c70 Ticket #525: Crash on call update or re-invite (Thanks Alexey) by Benny Prijono · 16 years ago
- 7d60d05 Ticket #504: final installment to support stereo audio all the way in PJMEDIA. Please see tickiet #504 for more info by Benny Prijono · 16 years ago
- 573b78c More ticket #61: (after rolling back previously buggy patch) Fixed bug in pjsua-lib with SRTP. If call is hold and resumed, SRTP transports will use itself as the underlying transport by Benny Prijono · 17 years ago
- 68f9e4f More ticket #61: bug in pjsua-lib with SRTP. If call is hold and resumed, SRTP transports will use itself as the underlying transport by Benny Prijono · 17 years ago
- 5297af9 Related to ticket #353: still memory leak with pjsua wav player (thanks Arie Velthoen) by Benny Prijono · 17 years ago
- 734fc2d More ticket #479: bug in pjmedia_transport_get_info(), the info should be initialized by caller by Benny Prijono · 17 years ago
- 7ffd775 Ticket #507: committed and tested g722 patch on Windows by Benny Prijono · 17 years ago
- 6e7c5ad More ticket #504: buffer overflow in splitcomb when handling stereo audio by Benny Prijono · 17 years ago
- e1a5a85 Ticket #479: allow media transport framework to return transport specific info (for example, to know whether SRTP is enabled) by Benny Prijono · 17 years ago
- 50f19b3 More ticket #495: bug in snd_clock_rate causing unability to open sound device on WinCE by Benny Prijono · 17 years ago
- f3758ee Ticket #495: ability to specify different clock rate when opening sound device by Benny Prijono · 17 years ago
- 2dbed82 Ticket #467: fixed issues with RTP/AVP vs RTP/SAVP negotiation by Benny Prijono · 17 years ago
- fc13bf6 Ticket #489: New PJSUA callbacks to notify application when media stream is created and destroyed by Benny Prijono · 17 years ago
- 0c06826 Added link to discussions about on_dtmf_callback() concurrency by Benny Prijono · 17 years ago
- 8b22ce1 Minor error: wrong logging info when printing RTP socket address by Benny Prijono · 17 years ago
- 25b2ea1 Return 406/Not Acceptable if SRTP negotiation failed instead of 500 by Benny Prijono · 17 years ago
- d817965 Ticket #61: Implement SRTP support in PJMEDIA and PJSUA-LIB, and updated applications because of the changes. This is a major modification back ported from SRTP branch. See ticket #61 for changelog detail of this commit by Benny Prijono · 17 years ago
- be41d86 Minor correction about WAV player parameter name and its comment in pjsua.h by Benny Prijono · 17 years ago
- 38fb3ea Related to ticket #437: optimize the stack usage of pjsua-lib by Benny Prijono · 17 years ago
- 91e567e Ticket #433: Failure in media negotiation when SDP contains two audio media lines (thanks Arie Velthoen) by Benny Prijono · 17 years ago
- 42d08d2 Related to Ticket #429: when bind address is specified and public address is not, the bind address should be used as the public address (thanks Arie Velthoen) by Benny Prijono · 17 years ago
- 0cdcd3f Ticket #429: Failed to create RTP/RTCP sockets when explicit bind address is specified (thanks Arie Velthoen) by Benny Prijono · 17 years ago
- fe0b1d6 Fixed mapped_address's address family not set causing assertion by Benny Prijono · 17 years ago
- 5186eae Ticket #420: updated pjmedia SDP and media UDP transport to support IPv6 by Benny Prijono · 17 years ago
- 23674a3 Ticket #421: initial IPv6 support: UDP transport by Benny Prijono · 17 years ago
- 8147f40 Ticket #14: Don't change RTP/RTCP SSRC on re-INVITE by Benny Prijono · 17 years ago
- 80eee89 Removed pjsua requirement to have consecutive RTCP mapped ports, and instead just print log message if mapped RTCP port is not adjacent to mapped RTP port by Benny Prijono · 17 years ago
- 6ba8c54 More ticket #399: added callback to report NAT detection result, and sends NAT type in SDP by Benny Prijono · 17 years ago
- cf0b4b2 Ticket #393: Added configuration to set basic audio frame length to minimize audio latency in pjsua-lib by Benny Prijono · 17 years ago