1. 7dd8659 SipSessionGroup: generating 32-bit random number for tag. by Hung-ying Tyan · 14 years ago
  2. 83dc899 SipSessionImpl: add MinExpiresHeader check. by Hung-ying Tyan · 14 years ago
  3. 8e188c7 SipSessionImpl: don't end call when an error occurs during a call. by Hung-ying Tyan · 14 years ago
  4. b6f7820 Merge "SipAudioCallImpl: deliver call change failure and don't end call when getting an error in a call." by Hung-ying Tyan · 14 years ago
  5. e2aa71b Merge "SIP telephony: don't end the call when getting error in a call." by Hung-ying Tyan · 14 years ago
  6. b4027bf Merge "SIP: demo call UI: hold call in onPause() and unhold in onResume() to make it work in simultaneous calls." by Hung-ying Tyan · 14 years ago
  7. 558dbba Use SIP OPTIONS instead of EMPTY message for keep-alive. by Chung-yih Wang · 14 years ago
  8. 880d1e3 SipAudioCallImpl: deliver call change failure and don't end call when getting an error in a call. by Hung-ying Tyan · 14 years ago
  9. 48a49de SIP telephony: don't end the call when getting error in a call. by Hung-ying Tyan · 14 years ago
  10. 2326290 SIP: demo call UI: hold call in onPause() and unhold in onResume() to make it work in simultaneous calls. by Hung-ying Tyan · 14 years ago
  11. e5545a6 Merge changes I9adc67d2,I32dd22af by Chung-yih Wang · 14 years ago
  12. 5854d60 SIP: fix a recursion bug when local IP becomes invalid (network disconnected). by Hung-ying Tyan · 14 years ago
  13. 4b7ae88 SIP telephony: integrate with new RTP stack and other fixes. by Hung-ying Tyan · 14 years ago
  14. 9e08977 RTP: use safer frame count to create AudioTrack and AudioRecord. by Chia-chi Yeh · 14 years ago
  15. 6045bde RTP: drain DeviceSocket before starting DeviceThread. by Chia-chi Yeh · 14 years ago
  16. 2be81a5 SIP telephony: add holding/swapping-calls, call-waiting by Hung-ying Tyan · 14 years ago
  17. 22d37ac RTP: temporarily make it froyo compatible to ease the development. by Chia-chi Yeh · 14 years ago
  18. 8092de3 RTP: tweak the lower bound of buffer size. by Chia-chi Yeh · 14 years ago
  19. de6a4ad RTP: add missing string.h to RtpStream.cpp. by Chia-chi Yeh · 14 years ago
  20. 92074f8 RTP: remove trailing spaces and add few logs. by Chia-chi Yeh · 14 years ago
  21. 6f70982 RTP: add the missing file for librtp_jni. by Chia-chi Yeh · 14 years ago
  22. 3e79283 RTP: add Java AudioGroup. by Chia-chi Yeh · 14 years ago
  23. 76e4dcf RTP: move AudioCodec to android.net.rtp. by Chia-chi Yeh · 14 years ago
  24. b02696a RTP: refactor out the network part from AudioStream to RtpStream. by Chia-chi Yeh · 14 years ago
  25. 7be5d9a RTP: add glue code for jni part. by Chia-chi Yeh · 14 years ago
  26. edf4a81 RTP: add AudioGroup which handles conference call, jitter buffer, and more. by Chia-chi Yeh · 14 years ago
  27. e5f71c2 RTP: abstract the network part from AudioStream to RtpStream. by Chia-chi Yeh · 14 years ago
  28. 3837b66 RTP: refactor native audio codec. by Chia-chi Yeh · 14 years ago
  29. ce350b8 SIP: cross out password when a profile is added to SipService by Hung-ying Tyan · 14 years ago
  30. 3732300 SipAudioCallImpl: revert the changes to hold/unhold implementation. by Hung-ying Tyan · 14 years ago
  31. b8ae93b SIP telephony: single call works (both incoming and outgoing). by Hung-ying Tyan · 14 years ago
  32. 599057e SipAudioCall: re-implemented holding/unholding a call. by Hung-ying Tyan · 14 years ago
  33. c1f79d2 SIP: add call busy handling to demo in-call screen by Hung-ying Tyan · 14 years ago
  34. d74c043 SipAudioCall: add new setListener() to explicitly specify immediate callback by Hung-ying Tyan · 14 years ago
  35. e6a9cc5 SIP: fix two bugs. by Hung-ying Tyan · 14 years ago
  36. 4aa0cef SIP telephony: add receiving call support (roughly) by Hung-ying Tyan · 14 years ago
  37. 701b995 SIP telephony: work-in-progres by Hung-ying Tyan · 14 years ago
  38. f7f6719 SIP telephony: mv SipPhoneFactory to where it should be. by Hung-ying Tyan · 14 years ago
  39. fe68ef2 SIP: work-in-progress for telephony integration. by Hung-ying Tyan · 14 years ago
  40. fcece04 SIP: duplicate PhoneApp for telephony integration development by Hung-ying Tyan · 14 years ago
  41. 5870f8d ISipService: add new open(), open3(), getListOfProfiles() by Hung-ying Tyan · 14 years ago
  42. 73f8437 SIP: change copyright year by Hung-ying Tyan · 14 years ago
  43. 9ea6739 SIP: duplicate PhoneApp for telephony integration development by Hung-ying Tyan · 14 years ago
  44. ff0fe66 SIP: first check-in of SipPhone and related classes. by Hung-ying Tyan · 14 years ago
  45. 831f9ac SIP: duplicate PhoneApp for telephony integration development by Hung-ying Tyan · 14 years ago
  46. 01d6af3 SIP: add sendDtmf() with callback by Hung-ying Tyan · 14 years ago
  47. 9562310 Add the missing resource file. by Chung-yih Wang · 14 years ago
  48. 8d001cf SIP: rearrange src files to separate settings and demo from framework code by Hung-ying Tyan · 14 years ago
  49. 729ff6b Some sip setting changes and registration fix. by Chung-yih Wang · 14 years ago
  50. aeeac7a SipCallUi: enable speaker and end-call buttons when making call by Hung-ying Tyan · 14 years ago
  51. 1c82627 SIP: SipAudioCallImpl: make ringback tone STREAM_VOICE_CALL by Hung-ying Tyan · 14 years ago
  52. 03f09c9 SIP: fixing SipServiceImpl by Hung-ying Tyan · 14 years ago
  53. b8ee2b9 SIP: WakeupTimer: fix timeout execution by Hung-ying Tyan · 14 years ago
  54. 13b3993 SIP: fix bugs in SipAudioCallImpl by Hung-ying Tyan · 14 years ago
  55. 3569d33 SIP: SipCallUi: synchronized on callbacks to prevent deadlock. by Hung-ying Tyan · 14 years ago
  56. d512aa6 SIP: WakeupTimer: check if the event queue is empty before retrieving first event by Hung-ying Tyan · 14 years ago
  57. 786d577 SIP: stop keepalive process when registration goes wrong by Hung-ying Tyan · 14 years ago
  58. 61d63bd SIP: fix bugs in WakeupTimer by Hung-ying Tyan · 14 years ago
  59. cb94c79 Add the call record for incoming call. by Chung-yih Wang · 14 years ago
  60. e505fe5 Fix the build break caused by the mutex initilization. by Chung-yih Wang · 14 years ago
  61. f0e64d1 Switch to pthread_mutex since the atomic framework is not finalized yet. by Chung-yih Wang · 14 years ago
  62. c4ccf8f SIP: acquire wifi lock when wifi is on by Hung-ying Tyan · 14 years ago
  63. 6b7f3df SIP: fix connectivity change handling when switching from a WIFI AP to another by Hung-ying Tyan · 14 years ago
  64. 0e9dd3d SIP: use listener proxy to safely call back in AutoRegistrationProcess. by Hung-ying Tyan · 14 years ago
  65. 23f4252 SIP: redesign WakeupTimer to align events better by Hung-ying Tyan · 14 years ago
  66. 5956418 Merge "SIP: remove WakeupTimer interface as it is used by SipServiceImpl with service context" by Hung-ying Tyan · 14 years ago
  67. 2c0c45f SIP: remove WakeupTimer interface as it is used by SipServiceImpl with service context by Hung-ying Tyan · 14 years ago
  68. a441001 Fix dtmf event bug. by Chung-yih Wang · 14 years ago
  69. 371e7e2 Fix the outbound proxy bug for sending keep-alive messages. by Chung-yih Wang · 14 years ago
  70. 74c6bab Add 'send keep-alive' feature in SipProfile. by Chung-yih Wang · 14 years ago
  71. cdf4554 SIP: finish call UI when a call ends. by Hung-ying Tyan · 14 years ago
  72. 6648c71 SIP: add FLog by Hung-ying Tyan · 14 years ago
  73. 169efa5 SIP: fix SipServiceImpl.createGroup(profile) by Hung-ying Tyan · 14 years ago
  74. 2c919db SIP: remove sendKeepAlive from ISipSession by Hung-ying Tyan · 14 years ago
  75. 13fb0d0 Merge "Send keepalive message periodically if the device is behind NAT." by Chung-yih Wang · 14 years ago
  76. 849a35a Send keepalive message periodically if the device is behind NAT. by Chung-yih Wang · 14 years ago
  77. 8ebabd1 SIP: fix speaker mode in SipAudioCallImpl and use toggle buttons in demo app for mute, hold and speaker mode. by Hung-ying Tyan · 14 years ago
  78. ce829d1 SipProfile: throw nullpointerexception early by Hung-ying Tyan · 14 years ago
  79. 6eb5145 Fix the exception in adding call log for an incoming call. by Chung-yih Wang · 14 years ago
  80. 38f0537 Add outgoing call receiver. by Chung-yih Wang · 14 years ago
  81. 7b8a6ad SIP: change copyright year from 2009 to 2010 by Hung-ying Tyan · 14 years ago
  82. 28a2484 Merge "SIP demo app: add proximity sensoring in call UI" by Hung-ying Tyan · 14 years ago
  83. 4a7dbf0 Merge "SIP demo app: add caller selection listener" by Hung-ying Tyan · 14 years ago
  84. 7728ba0 SIP demo app: add proximity sensoring in call UI by Hung-ying Tyan · 14 years ago
  85. d6c4c44 Add mic. gain for amplifying the outgoing voice stream. by Chung-yih Wang · 14 years ago
  86. c56d6f5 SIP demo app: add caller selection listener by Hung-ying Tyan · 14 years ago
  87. 0abc033 SIP: add demo call setup UI by Hung-ying Tyan · 14 years ago
  88. f22ce54 SIP: add demo in-call UI by Hung-ying Tyan · 14 years ago
  89. cacda01 SIP: minor cleanup in SipAudioCallImpl by Hung-ying Tyan · 14 years ago
  90. eadf8c3 Merge "Fix the invite ok issue for incoming call in sip channel with tcp transport." by Hung-ying Tyan · 14 years ago
  91. f7a6c82 Fix the invite ok issue for incoming call in sip channel with tcp transport. by Chung-yih Wang · 14 years ago
  92. 81d867c SipSettings: check isOpened() instead of isRegistered when creating menu by Hung-ying Tyan · 14 years ago
  93. ed85948 SipService: remove SipStack when close by Hung-ying Tyan · 14 years ago
  94. b10d267 SIP: add transport to SIP URI by Hung-ying Tyan · 14 years ago
  95. fc11383 SIP: add javadoc to android.net.sip classes by Hung-ying Tyan · 14 years ago
  96. 0f398eb Merge "Enable the TCP transport for SIP channel." by Chung-yih Wang · 14 years ago
  97. 0c5bdd9 Enable the TCP transport for SIP channel. by Chung-yih Wang · 14 years ago
  98. ebbf6da SIP: fix a time conversion bug in SipServiceImpl by Hung-ying Tyan · 14 years ago
  99. cff6f9c SIP: add space between entry name and warning message by Hung-ying Tyan · 14 years ago
  100. b52f45e SIP: make SipManager non-static to be consistent with other system services and add javadoc by Hung-ying Tyan · 14 years ago