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review.jami.net / pjproject / 879ad1a1b638e06e59ed9afd54e15d3da4b52f6c / pjsip
  1. 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
  2. 617b860 Fixed crash in SRTP when incoming SDP is received without any m= line (thanks Atik) by Benny Prijono · 16 years ago
  3. 7fff9f9 Ticket #522: Enable keep-alive for UDP transport even when STUN is not configured by Benny Prijono · 16 years ago
  4. 754a4df Ticket #521: Duplicate Authorization header when PJSIP is configured to send empty Authorization header (thanks Roland Klabunde) by Benny Prijono · 16 years ago
  5. c295d9f More ticket #514: bug in previous fix caused invite session to unable to respond to authentication in re-INVITE by Benny Prijono · 16 years ago
  6. 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
  7. 9ae5dfc More ticket #514: the last fix causes invite session to refuse to send CANCEL! by Benny Prijono · 16 years ago
  8. e8554ef Ticket #515 (Update Contact header in REGISTER for TCP/TLS transport) by Benny Prijono · 17 years ago
  9. 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
  10. 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
  11. 22e48c9 Ticket #514: bug with handling simultaneous outgoing re-INVITE, the invite session does not check if we have an ongoing INVITE transaction (thanks Philippe Leuba) by Benny Prijono · 17 years ago
  12. 5297af9 Related to ticket #353: still memory leak with pjsua wav player (thanks Arie Velthoen) by Benny Prijono · 17 years ago
  13. 734fc2d More ticket #479: bug in pjmedia_transport_get_info(), the info should be initialized by caller by Benny Prijono · 17 years ago
  14. 7ffd775 Ticket #507: committed and tested g722 patch on Windows by Benny Prijono · 17 years ago
  15. 6e7c5ad More ticket #504: buffer overflow in splitcomb when handling stereo audio by Benny Prijono · 17 years ago
  16. 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
  17. bcc7d0d More ticket #7: fixed undefined symbol on Symbian by Benny Prijono · 17 years ago
  18. 50f19b3 More ticket #495: bug in snd_clock_rate causing unability to open sound device on WinCE by Benny Prijono · 17 years ago
  19. 9963998 Ticket 467: dont send SDP with BYE! by Benny Prijono · 17 years ago
  20. 52cde92 Ticket #498: Option in client registration to ignore Contact address in REGISTER response by Benny Prijono · 17 years ago
  21. f3758ee Ticket #495: ability to specify different clock rate when opening sound device by Benny Prijono · 17 years ago
  22. 5d177b8 Fixed bug in ticket #455 in round-robin call ID allocation (thanks Truong Thanh Quang) by Benny Prijono · 17 years ago
  23. 4768c3c Ticket #7: Move PJSIP compile time configurations/settings (such as T1, T2 timers) to run-time (thanks Philippe Leuba) by Benny Prijono · 17 years ago
  24. 5933e05 Ticket #491: Crash in TCP/TLS transport when the listener is destroyed by Benny Prijono · 17 years ago
  25. 7433446 Ticket #492: Bug in strict route processing when challenged with 401/407 response by Benny Prijono · 17 years ago
  26. 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
  27. 8740238 Ticket #488: When outgoing request within dialog is responded with 481 or 408, should send BYE after terminating dialog (thanks Philippe Leuba) by Benny Prijono · 17 years ago
  28. 2dbed82 Ticket #467: fixed issues with RTP/AVP vs RTP/SAVP negotiation by Benny Prijono · 17 years ago
  29. fc13bf6 Ticket #489: New PJSUA callbacks to notify application when media stream is created and destroyed by Benny Prijono · 17 years ago
  30. 19450be Ticket #482: TCP keep-alive packets are corrupting SIP message (thanks Helmut Wolf) by Benny Prijono · 17 years ago
  31. 0ee4dde Ticket #481: Default TLS version should be TLSv1 (thanks Klaus Darilion) by Benny Prijono · 17 years ago
  32. 0c06826 Added link to discussions about on_dtmf_callback() concurrency by Benny Prijono · 17 years ago
  33. 8b22ce1 Minor error: wrong logging info when printing RTP socket address by Benny Prijono · 17 years ago
  34. db844a4 More ticket #61: fixed signaling security level calculation for SRTP by Benny Prijono · 17 years ago
  35. bc2219b Added pj_strstr() and pj_stristr() in pjlib by Benny Prijono · 17 years ago
  36. 423f641 Bug: source address not initialized in loop transport causing pjsip-test to fail by Benny Prijono · 17 years ago
  37. a7b376b Fixed doxygen comments everywhere by Benny Prijono · 17 years ago
  38. 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
  39. 25b2ea1 Return 406/Not Acceptable if SRTP negotiation failed instead of 500 by Benny Prijono · 17 years ago
  40. 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
  41. 87a9021 Related to ticket #61: added new invite session API pjsip_inv_verify_request() which takes additional remote SDP, to avoid parsing SDP multiple times by Benny Prijono · 17 years ago
  42. 5b221d4 Related to ticket #61: Added PJSIP_ESESSIONINSECURE error code in sip_errno to require that secure session shall be used (needed by SRTP) by Benny Prijono · 17 years ago
  43. 5773cd6 Ticket #455: allocate pjsua call id in round robin fashion by Benny Prijono · 17 years ago
  44. 4190cf9 Ticket #453: Log level is not set in PJSUA-LIB (thanks Simon Farmer) by Benny Prijono · 17 years ago
  45. be41d86 Minor correction about WAV player parameter name and its comment in pjsua.h by Benny Prijono · 17 years ago
  46. 58add7c Minor correction about WAV player parameter name and its comment in pjsua.h by Benny Prijono · 17 years ago
  47. 0b41aa5 Fixed compilation warnings on OSX 10.5 by Benny Prijono · 17 years ago
  48. 59b3aed Reply with 488+SDP instead of 415 when incoming SDP is not acceptable (thanks Alain Totouom) by Benny Prijono · 17 years ago
  49. 37c710b Added PJSUA_DEFAULT_AUDIO_FRAME_PTIME setting and changed default iLBC mode from 20 to 30 by Benny Prijono · 17 years ago
  50. e723b92 Ticket #443: Overflow in dump_media_session() (thanks Simon Farmer) by Benny Prijono · 17 years ago
  51. 1402a4a Protect against division by zero in pjsua's dump_media_session (thanks Simon Farmer) by Benny Prijono · 17 years ago
  52. 38fb3ea Related to ticket #437: optimize the stack usage of pjsua-lib by Benny Prijono · 17 years ago
  53. 91e567e Ticket #433: Failure in media negotiation when SDP contains two audio media lines (thanks Arie Velthoen) by Benny Prijono · 17 years ago
  54. 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
  55. 9f6c90c Reverted back changes in r1631 by Benny Prijono · 17 years ago
  56. ae63e55 Make pj_ioqueue_op_key_init() in pjsip_tx_data_create() more correct by Benny Prijono · 17 years ago
  57. 3a46283 Fixed case when tdata is NULL pjsip_tx_data_get_info() by Benny Prijono · 17 years ago
  58. 0cdcd3f Ticket #429: Failed to create RTP/RTCP sockets when explicit bind address is specified (thanks Arie Velthoen) by Benny Prijono · 17 years ago
  59. db4eb81 IPv6 logging in account keep-alive in pjsua-lib by Benny Prijono · 17 years ago
  60. 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
  61. b247714 Fixed error when creating TLS transport in pjsua-lib (the TLS type was misidentified was UDP) by Benny Prijono · 17 years ago
  62. 19b2937 Fixed Contact generation failure and crash on unregistration if regc initialization failed by Benny Prijono · 17 years ago
  63. fe0b1d6 Fixed mapped_address's address family not set causing assertion by Benny Prijono · 17 years ago
  64. 5186eae Ticket #420: updated pjmedia SDP and media UDP transport to support IPv6 by Benny Prijono · 17 years ago
  65. 7ebdb3e Fixed compilation errors/warnings on Linux with the recent IPv6 commits 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. fe7d87b Fixed difference in the declaration of log callback between pjsua-lib and pjlib (thanks Arie Velthoen) by Benny Prijono · 17 years ago
  69. d5f9f42 Ticket #416: Allow application to handle sending ACK manually by Benny Prijono · 17 years ago
  70. c514576 Fixed ticket #426: Respond incoming CANCEL with no matching INVITE with 481 (thanks Sergey Bakulin) by Benny Prijono · 17 years ago
  71. 99cd029 Ticket #424: Added API to retrieve number of transactions and dialogs (thanks Sergey Bakulin) by Benny Prijono · 17 years ago
  72. 37db51c Fixed ticket #423: Client registration (pjsip_regc) doesn't obey explicit transport selection (thanks Hitesh) by Benny Prijono · 17 years ago
  73. 8147f40 Ticket #14: Don't change RTP/RTCP SSRC on re-INVITE by Benny Prijono · 17 years ago
  74. d07f716 Fixed wrong doxygen comment on various SIP parsing function, about null-terminating the input text by Benny Prijono · 17 years ago
  75. da9080f Fixed milenage build on eVC etc. by Benny Prijono · 17 years ago
  76. 744aca0 Do not resolve SIP address with STUN if public address is configured by Benny Prijono · 17 years ago
  77. bc9baa9 Enable AKA support in the build by default by Benny Prijono · 17 years ago
  78. f0f8fd1 Fixed compilation warnings/errors under C++ mode by Benny Prijono · 17 years ago
  79. 1d699e8 Deregister PJSIP_SYN_ERR_EXCEPTION upon destroying SIP endpoint to allow restarting SIP endpoint without shutting down pjlib (thanks Phil Torre) by Benny Prijono · 17 years ago
  80. 2fc98ba Update the digest AKAv2-MD5 implementation, we can now login to OpenIMSCore by Benny Prijono · 17 years ago
  81. fc8bb14 Remove SDP from 487 response when incoming INVITE comes without SDP and we are sending offer in 18x response by Benny Prijono · 17 years ago
  82. c5cbc05 Accept UPDATE without SDP (this is a valid scenario according to session timer RFC) by Benny Prijono · 17 years ago
  83. 8eb07bf In pjsua-lib, treat incoming SDP with a=sendonly as hold request, and respond with a=inactive by Benny Prijono · 17 years ago
  84. 48ab2b7 - Added option to send empty Authorization header in outgoing requests by Benny Prijono · 17 years ago
  85. 3f302ff Add Contact header to UPDATE request within dialog by Benny Prijono · 17 years ago
  86. 94b2244 Allow zero in outgoing CSeq in request creation within dialog by Benny Prijono · 17 years ago
  87. 32767ec Change default PUBLISH interval from 60 seconds to PJSUA_PRES_TIMER by Benny Prijono · 17 years ago
  88. 2fcca5c Fixed bug in route set calculation: prevent updating route set upon receiving failure response (e.g. 401/407 response), and remove the first_cseq check since this would not work when the first request is challenged by Benny Prijono · 17 years ago
  89. c1b1c0a Validate SDP in incoming message before passing it to negotiator (otherwise assertion will occur if SDP contains error) by Benny Prijono · 17 years ago
  90. ea9fd39 The NAT type investigation in incoming INVITE caused assertion error when the INVITE comes without SDP by Benny Prijono · 17 years ago
  91. 5e2f683 SIP TCP and TLS transports may call send completion callback with bytes_set=0 when send operation fails because outgoing connection is cancelled (for example, application quits). This will trigger assertion error in transaction because transaction only expects positive or negative bytes_set number, but not zero by Benny Prijono · 17 years ago
  92. 8dbd863 SIP TCP and TLS transports may call send completion callback with bytes_set=0 when send operation fails because outgoing connection is cancelled (for example, application quits). This will trigger assertion error in transaction because transaction only expects positive or negative bytes_set number, but not zero by Benny Prijono · 17 years ago
  93. 42307c7 Update ticket #408: although route set must not be updated on subsequent requests, dialog MUST recompute the route set upon receiving 2xx response if the route set was previously computed from 1xx response by Benny Prijono · 17 years ago
  94. 7129cc7 Increment SDP version upon creating subsequent offer inside the call (ref: Sipit21/Mon/12:30) by Benny Prijono · 17 years ago
  95. e1c984f Fixed bug with detecting successful unregistration request. Previously, successful unregistration was mistakenly treated as successful registration when it contains no Contact header and has positive Expires header value by Benny Prijono · 17 years ago
  96. 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
  97. 2568c74 Changed Service-Route processing to append S-R to existing route set rather than overwriting them by Benny Prijono · 17 years ago
  98. 9761c4c Fixed wrong comment on audio_frame_ptime default value by Benny Prijono · 17 years ago
  99. efe0d32 Allow empty reason phrase in SIP responses during parsing (thanks Roman Puls) by Benny Prijono · 17 years ago
  100. 860be56 Ticket #410: Endless authentication retries when 401/407 response contains no challenge by Benny Prijono · 17 years ago