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gerrit-public.fairphone.software / platform / frameworks / base / 090a3dc33fbb0d450dd5823fd2756360e4acdd12 / voip
  1. f053292 Fix SIP bug of different transport/port used for requests. by Chung-yih Wang · 14 years ago
  2. e2abd10 Merge "Set AudioGroup mode according to audio settings" into gingerbread by Hung-ying Tyan · 14 years ago
  3. 06e8cdc Fix race between ending and answering a SIP call. by Hung-ying Tyan · 14 years ago
  4. fa81463 Set AudioGroup mode according to audio settings by Hung-ying Tyan · 14 years ago
  5. 4189d99 Do not suppress error feedback during a SIP call. by Hung-ying Tyan · 14 years ago
  6. 349f350 Merge "Correct SipService.isOpened() implementation." into gingerbread by Hung-ying Tyan · 14 years ago
  7. d9e1230 Merge "Notify SipSessions before closing SIP stack." into gingerbread by Hung-ying Tyan · 14 years ago
  8. 0bba953 Merge "Throw proper exceptions in SipManager" into gingerbread by Hung-ying Tyan · 14 years ago
  9. bd399b0 Merge "RTP: Pause echo suppressor when far-end volume is low." into gingerbread by Chia-chi Yeh · 14 years ago
  10. 8d1b2a1 Throw proper exceptions in SipManager by Hung-ying Tyan · 14 years ago
  11. 262cdfc Correct SipService.isOpened() implementation. by Hung-ying Tyan · 14 years ago
  12. 02b1d68 docs: revise javadocs for sip by Scott Main · 14 years ago
  13. 0c7d306 RTP: Pause echo suppressor when far-end volume is low. by Chia-chi Yeh · 14 years ago
  14. 5d0c5cf Notify SipSessions before closing SIP stack. by Hung-ying Tyan · 14 years ago
  15. 60c45d0 Clean up pending sessions on incoming call in SipService by Hung-ying Tyan · 14 years ago
  16. 703aae0 Merge "RTP: Fix non-zero DC in EchoSuppressor caused while aggregating samples." into gingerbread by Chia-chi Yeh · 14 years ago
  17. 8a68b52 RTP: Fix non-zero DC in EchoSuppressor caused while aggregating samples. by Chia-chi Yeh · 14 years ago
  18. e87b644 Add permission requirements to SipAudioCall and SipManager javadoc. by Hung-ying Tyan · 14 years ago
  19. 9b449e5 Remove ringtone API from SipAudioCall. by Hung-ying Tyan · 14 years ago
  20. 4944fdd Periodically scan wifi when wifi is not connected and wifi lock is by Chung-yih Wang · 14 years ago
  21. 0a6e717 Handle dialing a SIP call to self. by Hung-ying Tyan · 14 years ago
  22. 431bb22 Reduce logging. by Joe Onorato · 14 years ago
  23. b4116c0 Fix the incorrect environment variable name for the thread pool size. by Chung-yih Wang · 14 years ago
  24. 45bd830 Merge "Uncomment SIP/VOIP feature check in SipManager." into gingerbread by John Huang · 14 years ago
  25. a0cdfbf Uncomment SIP/VOIP feature check in SipManager. by Hung-ying Tyan · 14 years ago
  26. 66cc535 Set the thread pool size of NIST sip stack to one. by Chung-yih Wang · 14 years ago
  27. 28f63c0 SipService: add wake lock for incoming INVITE packets. by Hung-ying Tyan · 14 years ago
  28. bd57eea SipService: add wake lock for multiple components. by Hung-ying Tyan · 14 years ago
  29. 4f8fd10 Make SipService listen to WIFI state change events. by Hung-ying Tyan · 14 years ago
  30. f1b1eec Merge "SipService: mScreenOn is flipped to wrong value." into gingerbread by Hung-ying Tyan · 14 years ago
  31. d6fc979 SipService: mScreenOn is flipped to wrong value. by Hung-ying Tyan · 14 years ago
  32. ebc886c Fix SipSessionGroup from throwing ConcurrentModificationException by Hung-ying Tyan · 14 years ago
  33. 685b61b SipService: fix a missing switch-case break. by Hung-ying Tyan · 14 years ago
  34. 692cac9 SipHelper: add debug log for challenge responses. by Hung-ying Tyan · 14 years ago
  35. c7fda18 Do not release the wifi lock if the screen is off. by Chung-yih Wang · 14 years ago
  36. aa562ff SipService: add permission check for using API by Hung-ying Tyan · 14 years ago
  37. 08faac3 Unhide SIP API. by Hung-ying Tyan · 14 years ago
  38. 4483232 Suppress harder for echo without affecting the volume of real voice. by Chung-yih Wang · 14 years ago
  39. 9db99a4 Make SipService broadcast SIP_SERVICE_UP when it's up. by Hung-ying Tyan · 14 years ago
  40. 16c29bd Merge "SIP: Fix busy authentication loop." into gingerbread by Hung-ying Tyan · 14 years ago
  41. ee8a884 SIP: Fix busy authentication loop. by Hung-ying Tyan · 14 years ago
  42. fb116fb Misc fixes for sim-eng build. by Chung-yih Wang · 14 years ago
  43. 718e003 Merge "SIP: add SERVER_UNREACHABLE error code." into gingerbread by Hung-ying Tyan · 14 years ago
  44. c6548fd9 SIP: add SERVER_UNREACHABLE error code. by Hung-ying Tyan · 14 years ago
  45. 323d367 SipService: supply PendingIntent when open a profile. by Hung-ying Tyan · 14 years ago
  46. 4fc04f1 Merge "RTP: Add a baseline echo suppressor." into gingerbread by Chia-chi Yeh · 14 years ago
  47. a8a1009 RTP: Add a baseline echo suppressor. by Chia-chi Yeh · 14 years ago
  48. 9ea96c6 SIP: minor fixes. by Hung-ying Tyan · 14 years ago
  49. 274e3b5 Merge "RTP: Start AudioRecord before AudioTrack to avoid being disabled." into gingerbread by Chia-chi Yeh · 14 years ago
  50. 063d02b Merge "SipService: turn off verbose logging" into gingerbread by Hung-ying Tyan · 14 years ago
  51. 67ecb5b RTP: Start AudioRecord before AudioTrack to avoid being disabled. by Chia-chi Yeh · 14 years ago
  52. b031957 SipService: turn off verbose logging by Hung-ying Tyan · 14 years ago
  53. 21ae1ad RTP: Minor fixes with polishing. by Chia-chi Yeh · 14 years ago
  54. d29e075 Merge "Add uri field to SipManager.ListenerRelay" into gingerbread by Hung-ying Tyan · 14 years ago
  55. 9e1d308 Add uri field to SipManager.ListenerRelay by Hung-ying Tyan · 14 years ago
  56. dfd1484 Merge "RTP: Adjust the jitter buffer to 512ms." into gingerbread by Chia-chi Yeh · 14 years ago
  57. 3520bd4 RTP: Adjust the jitter buffer to 512ms. by Chia-chi Yeh · 14 years ago
  58. 6a53489 SipService: add UID check. by Hung-ying Tyan · 14 years ago
  59. 0a537b7 Merge "RTP: Enable AMR codec." into gingerbread by Chia-chi Yeh · 14 years ago
  60. 2365b78 Merge "SIP: misc fixes." into gingerbread by Hung-ying Tyan · 14 years ago
  61. f88fc1f RTP: Enable AMR codec. by Chia-chi Yeh · 14 years ago
  62. fb3a98b SIP: misc fixes. by Hung-ying Tyan · 14 years ago
  63. f4ae942 RTP: Enable GSM-EFR codec. by Chia-chi Yeh · 14 years ago
  64. fe52989 RTP: Revise the workaround of private addresses and fix bugs. by Chia-chi Yeh · 14 years ago
  65. e006e4d Merge changes Iae1913fb,I38dbefef into gingerbread by Chia-chi Yeh · 14 years ago
  66. a6f950c RTP: Enable GSM codec. by Chia-chi Yeh · 14 years ago
  67. 78c11b3 RTP: Refactor out G711 codecs into another file. by Chia-chi Yeh · 14 years ago
  68. 320cdcb Merge "RTP: Delay the initialization of AudioTrack and AudioRecord." into gingerbread by Chia-chi Yeh · 14 years ago
  69. 9083c84 RTP: Delay the initialization of AudioTrack and AudioRecord. by Chia-chi Yeh · 14 years ago
  70. 6057cd0 SIP: Feedback any provisional responses in addition to RING by Hung-ying Tyan · 14 years ago
  71. 624d5b4 SIP: add DisconnectCause.SERVER_ERROR by Hung-ying Tyan · 14 years ago
  72. 7e54ef7 Move SipService out of SystemServer to phone process. by Hung-ying Tyan · 14 years ago
  73. fd144d7 Merge "SipAudioCall: remove SipManager dependency." into gingerbread by Hung-ying Tyan · 14 years ago
  74. 00a2206 SipService: handle cross-domain authentication error by Hung-ying Tyan · 14 years ago
  75. bd22942 Fix the unhold issue especially if one is behind NAT. by Chung-yih Wang · 14 years ago
  76. 3a4197e SipAudioCall: remove SipManager dependency. by Hung-ying Tyan · 14 years ago
  77. 658bec9 SDP: remove dead code. by Chia-chi Yeh · 14 years ago
  78. 84a357b Refactoring SIP classes to get ready for API review. by Hung-ying Tyan · 14 years ago
  79. 0b7d6de Fix the build. by repo sync · 14 years ago
  80. 84f7f6b SIP: Make SipAudioCallImpl use SimpleSessionDescription instead of javax.sdp. by repo sync · 14 years ago
  81. e6c0c10 SDP: Add a simple class to help manipulate session descriptions. by Chia-chi Yeh · 14 years ago
  82. 7a69aef RTP: Add log throttle for "no data". by repo sync · 14 years ago
  83. 4033a67 RTP: Update native part to reflect the API change. by Chia-chi Yeh · 14 years ago
  84. 37adc52 RTP: Add two getters to retrieve the current configuration from AudioStream. by Chia-chi Yeh · 14 years ago
  85. 32e106b RTP: Extend codec capability and update the APIs. by Chia-chi Yeh · 14 years ago
  86. 8544560 SipPhone: fix missing-call DisconnectCause feedback by Hung-ying Tyan · 14 years ago
  87. 9796379 SIP: convert enum to static final int. by Hung-ying Tyan · 14 years ago
  88. c4b8747 SIP: add config flag for wifi-only configuration. by Hung-ying Tyan · 14 years ago
  89. afa583e SipAudioCall: expose startAudio() by Hung-ying Tyan · 14 years ago
  90. 9352cf1 Add timer to SIP session creation process. by Hung-ying Tyan · 14 years ago
  91. 286bb5a Fix links in SIP API javadoc. by Hung-ying Tyan · 14 years ago
  92. ae076d3 SIP: add PEER_NOT_REACHABLE error feedback. by Hung-ying Tyan · 14 years ago
  93. 12bec5d SipService: ignore connect event for non-active networks. by Hung-ying Tyan · 14 years ago
  94. 13f6270 SipAudioCall: use SipErrorCode instead of string in onError() by Hung-ying Tyan · 14 years ago
  95. 99bf4e4 SIP: remove dependency on javax.sip by Hung-ying Tyan · 14 years ago
  96. d231aa8 SipService: deliver connectivity change to all sessions. by Hung-ying Tyan · 14 years ago
  97. 3d7606a SIP: enhance timeout and registration status feedback. by Hung-ying Tyan · 14 years ago
  98. 25b52a2 SIP: remove dependency on javax.sip.SipException. by Hung-ying Tyan · 14 years ago
  99. 903e103 SIP: add SipErrorCode for error feedback. by Hung-ying Tyan · 14 years ago
  100. f6936a3a Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread by Chia-chi Yeh · 14 years ago