Gitiles
Code Review Sign In
review.jami.net / pjproject / ab8dba93e3817cc99487ec03fcb53dc882e36054 / pjsip / src / pjsua-lib
  1. ab8dba9 Added more Python tests: offer with multiple media lines by Benny Prijono · 16 years ago
  2. 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
  3. 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
  4. 2c484e4 Fixed assertion in invite session when INVITE has bad SDP because the SDP is given without having been validated first by Benny Prijono · 16 years ago
  5. 4375f90 Ticket #544: Fixed SRTP on hold+reinvite scenario by Nanang Izzuddin · 16 years ago
  6. 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
  7. 2331d20 Fixed bug: media got deactivated when SDP negotiation fails on re-INVITE or UPDATE while it should be unaffected by Benny Prijono · 16 years ago
  8. 32177c0 Large changeset to replace all occurences of year 2007 with 2008 in the copyright notice by Benny Prijono · 16 years ago
  9. 224b4e2 Ticket #549: major modification in media transport API to support more offer/answer scenarios by Benny Prijono · 16 years ago
  10. 148fd39 More ticket #540: updated snd-auto-close to work friendly with call by Nanang Izzuddin · 16 years ago
  11. 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
  12. 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
  13. fd461eb Added more validations & a new API (thanks Florian Bomers): by Nanang Izzuddin · 16 years ago
  14. f76e139 More ticket #485: added TURN support in PJSUA-LIB API by Benny Prijono · 16 years ago
  15. e85a183 Fixed bug wrong option for resample port between conference bridge and sound device by Nanang Izzuddin · 16 years ago
  16. 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
  17. 2d4ee7d More on ticket #535: updated files using and related to math.h by Nanang Izzuddin · 16 years ago
  18. 5516f91 Fixed bug: NOTIFY is sent continuously on PJSUA-LIB shutdown by Benny Prijono · 16 years ago
  19. 83088f3 Fixed bug in invalid Contact address being generated upon NAT detection, when no username part is present in the account ID by Benny Prijono · 16 years ago
  20. ddaaf6a Use the smart Contact header for TCP/TLS by Benny Prijono · 16 years ago
  21. 24a2185 Fixed miscellaneous compile warnings/errors when built with C++ mode by Benny Prijono · 16 years ago
  22. 53a7c70 Ticket #525: Crash on call update or re-invite (Thanks Alexey) by Benny Prijono · 16 years ago
  23. c54dcb3 As per list report, changed the default response to incoming REFER from 200 to 202 as some gateways do not like this. Thanks Pedro Sanchez for the report by Benny Prijono · 16 years ago
  24. 617b860 Fixed crash in SRTP when incoming SDP is received without any m= line (thanks Atik) by Benny Prijono · 16 years ago
  25. 7fff9f9 Ticket #522: Enable keep-alive for UDP transport even when STUN is not configured by Benny Prijono · 16 years ago
  26. 7d60d05 Ticket #504: final installment to support stereo audio all the way in PJMEDIA. Please see tickiet #504 for more info by Benny Prijono · 17 years ago
  27. e8554ef Ticket #515 (Update Contact header in REGISTER for TCP/TLS transport) by Benny Prijono · 17 years ago
  28. 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
  29. 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
  30. 5297af9 Related to ticket #353: still memory leak with pjsua wav player (thanks Arie Velthoen) by Benny Prijono · 17 years ago
  31. 734fc2d More ticket #479: bug in pjmedia_transport_get_info(), the info should be initialized by caller by Benny Prijono · 17 years ago
  32. 7ffd775 Ticket #507: committed and tested g722 patch on Windows by Benny Prijono · 17 years ago
  33. 6e7c5ad More ticket #504: buffer overflow in splitcomb when handling stereo audio by Benny Prijono · 17 years ago
  34. 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
  35. 50f19b3 More ticket #495: bug in snd_clock_rate causing unability to open sound device on WinCE by Benny Prijono · 17 years ago
  36. 9963998 Ticket 467: dont send SDP with BYE! by Benny Prijono · 17 years ago
  37. f3758ee Ticket #495: ability to specify different clock rate when opening sound device by Benny Prijono · 17 years ago
  38. 5d177b8 Fixed bug in ticket #455 in round-robin call ID allocation (thanks Truong Thanh Quang) by Benny Prijono · 17 years ago
  39. 8389c31 Ticket #412: increased randomness of guid_simple.c to 192-bits, and seed random number generator in pjsua_core.c by Benny Prijono · 17 years ago
  40. 2dbed82 Ticket #467: fixed issues with RTP/AVP vs RTP/SAVP negotiation by Benny Prijono · 17 years ago
  41. fc13bf6 Ticket #489: New PJSUA callbacks to notify application when media stream is created and destroyed by Benny Prijono · 17 years ago
  42. 0c06826 Added link to discussions about on_dtmf_callback() concurrency by Benny Prijono · 17 years ago
  43. 8b22ce1 Minor error: wrong logging info when printing RTP socket address by Benny Prijono · 17 years ago
  44. db844a4 More ticket #61: fixed signaling security level calculation for SRTP by Benny Prijono · 17 years ago
  45. bc2219b Added pj_strstr() and pj_stristr() in pjlib by Benny Prijono · 17 years ago
  46. f650898 More ticket #61: fix bug in secure signaling determination, and added --srtp-secure option in pjsua to control signaling security requirement for SRTP by Benny Prijono · 17 years ago
  47. 25b2ea1 Return 406/Not Acceptable if SRTP negotiation failed instead of 500 by Benny Prijono · 17 years ago
  48. 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
  49. 5773cd6 Ticket #455: allocate pjsua call id in round robin fashion by Benny Prijono · 17 years ago
  50. 4190cf9 Ticket #453: Log level is not set in PJSUA-LIB (thanks Simon Farmer) by Benny Prijono · 17 years ago
  51. be41d86 Minor correction about WAV player parameter name and its comment in pjsua.h by Benny Prijono · 17 years ago
  52. 59b3aed Reply with 488+SDP instead of 415 when incoming SDP is not acceptable (thanks Alain Totouom) by Benny Prijono · 17 years ago
  53. 37c710b Added PJSUA_DEFAULT_AUDIO_FRAME_PTIME setting and changed default iLBC mode from 20 to 30 by Benny Prijono · 17 years ago
  54. e723b92 Ticket #443: Overflow in dump_media_session() (thanks Simon Farmer) by Benny Prijono · 17 years ago
  55. 1402a4a Protect against division by zero in pjsua's dump_media_session (thanks Simon Farmer) by Benny Prijono · 17 years ago
  56. 38fb3ea Related to ticket #437: optimize the stack usage of pjsua-lib by Benny Prijono · 17 years ago
  57. 91e567e Ticket #433: Failure in media negotiation when SDP contains two audio media lines (thanks Arie Velthoen) by Benny Prijono · 17 years ago
  58. 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
  59. 0cdcd3f Ticket #429: Failed to create RTP/RTCP sockets when explicit bind address is specified (thanks Arie Velthoen) by Benny Prijono · 17 years ago
  60. db4eb81 IPv6 logging in account keep-alive in pjsua-lib by Benny Prijono · 17 years ago
  61. 5f55a1f Fixed run-time error on MacOS X (and possibly BSD based systems) when bind() is called with larger addrlen by Benny Prijono · 17 years ago
  62. b247714 Fixed error when creating TLS transport in pjsua-lib (the TLS type was misidentified was UDP) by Benny Prijono · 17 years ago
  63. 19b2937 Fixed Contact generation failure and crash on unregistration if regc initialization failed by Benny Prijono · 17 years ago
  64. fe0b1d6 Fixed mapped_address's address family not set causing assertion by Benny Prijono · 17 years ago
  65. 5186eae Ticket #420: updated pjmedia SDP and media UDP transport to support IPv6 by Benny Prijono · 17 years ago
  66. d0bd498 More ticket #421: fixed SIP messaging components to support IPv6 format by Benny Prijono · 17 years ago
  67. 23674a3 Ticket #421: initial IPv6 support: UDP transport by Benny Prijono · 17 years ago
  68. 8147f40 Ticket #14: Don't change RTP/RTCP SSRC on re-INVITE by Benny Prijono · 17 years ago
  69. 744aca0 Do not resolve SIP address with STUN if public address is configured by Benny Prijono · 17 years ago
  70. f0f8fd1 Fixed compilation warnings/errors under C++ mode by Benny Prijono · 17 years ago
  71. 8eb07bf In pjsua-lib, treat incoming SDP with a=sendonly as hold request, and respond with a=inactive by Benny Prijono · 17 years ago
  72. 48ab2b7 - Added option to send empty Authorization header in outgoing requests by Benny Prijono · 17 years ago
  73. 32767ec Change default PUBLISH interval from 60 seconds to PJSUA_PRES_TIMER by Benny Prijono · 17 years ago
  74. ea9fd39 The NAT type investigation in incoming INVITE caused assertion error when the INVITE comes without SDP by Benny Prijono · 17 years ago
  75. 7129cc7 Increment SDP version upon creating subsequent offer inside the call (ref: Sipit21/Mon/12:30) by Benny Prijono · 17 years ago
  76. 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
  77. 2568c74 Changed Service-Route processing to append S-R to existing route set rather than overwriting them by Benny Prijono · 17 years ago
  78. e083fd5 More ticket #385: 100rel support should be enabled by default by Benny Prijono · 17 years ago
  79. bddef2c Ticket #407: keep-alive for UDP transports in PJSUA-LIB by Benny Prijono · 17 years ago
  80. 1086143 Ticket #406: New PJSUA API to update buddy's presence subscription by Benny Prijono · 17 years ago
  81. a17496a Ticket #405: Subscribe to buddy presence upon receiving incoming subscription from the buddy by Benny Prijono · 17 years ago
  82. 91a6a17 More ticket #399: added PJSUA API to retrieve the remote NAT type by Benny Prijono · 17 years ago
  83. 44e88ea Added pjsua_get_var() to access pjsua_var from a DLL (thanks Tomas Valenta) by Benny Prijono · 17 years ago
  84. 2a67ea4 Continuing ticket #400: Only process Service-Route header if enable_service_route (--service-route option in pjsua) is set by Benny Prijono · 17 years ago
  85. f020ab2 Updated Service-Route calculation by Benny Prijono · 17 years ago
  86. a2a2d41 Ticket #400: initial support for Service-Route header, still untested by Benny Prijono · 17 years ago
  87. 6ba8c54 More ticket #399: added callback to report NAT detection result, and sends NAT type in SDP by Benny Prijono · 17 years ago
  88. 28f673a Continuing ticket #396: tested digest AKAv1, implemented AKAv2, and some works in the authentication framework to support it by Benny Prijono · 17 years ago
  89. 4ab9fbb Ticket #399: Initial implementation of tool to perform NAT type detection/classification by Benny Prijono · 17 years ago
  90. 7977f9f Ticket #396: initial implementation of digest AKA (akav1-md5) authentication for IMS/3GPP by Benny Prijono · 17 years ago
  91. cf0b4b2 Ticket #393: Added configuration to set basic audio frame length to minimize audio latency in pjsua-lib by Benny Prijono · 17 years ago
  92. feb69f4 Ticket #391: Added framework to send and receive arbitrary requests within call in PJSUA-LIB, with samples to send/receive DTMF with INFO in pjsua application by Benny Prijono · 17 years ago
  93. c08682e Ticket #389: Added new commands in pjsua to change codec priorities and send UPDATE by Benny Prijono · 17 years ago
  94. 7a686d0 Related to ticket #385: fixed bug in pjsua-lib: 100rel module is not initialized because PJSIP_HAS_100REL macro has been removed by Benny Prijono · 17 years ago
  95. 1f7767b Ticket 5: Support for SIP UPDATE (RFC 3311) and fix the offer/answer negotiation by Benny Prijono · 17 years ago
  96. a41d66e Fixed ticket #386: Over-deinitialize sound subsystem in pjsua_media.c (thanks Jiandong Ruan) by Benny Prijono · 17 years ago
  97. dcfc0ba Ticket #385: Support for reliable provisional response (100rel, PRACK) by Benny Prijono · 17 years ago
  98. 15b0230 Ticket #381: auto-update the IP address in Contact according to the address/port received in REGISTER response by Benny Prijono · 17 years ago
  99. 627cbb4 Exported some private pjsua_call.c functions related to call quality statistics by Benny Prijono · 17 years ago
  100. 11da9bc Implemented ticket #373: Packet loss simulation in PJMEDIA ICE transport by Benny Prijono · 17 years ago