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gerrit-public.fairphone.software / platform / frameworks / base / b9fedd9771a6aedf068155fd91f15baf52e668cd / voip
  1. 51d2ada am 1f34ffd7: am 5cab38ba: Merge "SIP: minor fixes." into gingerbread by Hung-ying Tyan · 14 years ago
  2. 9ea96c6 SIP: minor fixes. by Hung-ying Tyan · 14 years ago
  3. 82fb4ef am c38e6ae4: am 274e3b5d: Merge "RTP: Start AudioRecord before AudioTrack to avoid being disabled." into gingerbread by Chia-chi Yeh · 14 years ago
  4. 1041bb7 am 9af6b536: am 063d02bb: Merge "SipService: turn off verbose logging" into gingerbread by Hung-ying Tyan · 14 years ago
  5. 274e3b5 Merge "RTP: Start AudioRecord before AudioTrack to avoid being disabled." into gingerbread by Chia-chi Yeh · 14 years ago
  6. 063d02b Merge "SipService: turn off verbose logging" into gingerbread by Hung-ying Tyan · 14 years ago
  7. 67ecb5b RTP: Start AudioRecord before AudioTrack to avoid being disabled. by Chia-chi Yeh · 14 years ago
  8. b031957 SipService: turn off verbose logging by Hung-ying Tyan · 14 years ago
  9. 2e88d0c am 2b133fc0: am 21ae1ad6: RTP: Minor fixes with polishing. by Chia-chi Yeh · 14 years ago
  10. e0ed9db am c79e74ec: am d29e0754: Merge "Add uri field to SipManager.ListenerRelay" into gingerbread by Hung-ying Tyan · 14 years ago
  11. bf45f19 am f6381ec1: am dfd1484e: Merge "RTP: Adjust the jitter buffer to 512ms." into gingerbread by Chia-chi Yeh · 14 years ago
  12. f3da1ea am 34552149: am 6a53489a: SipService: add UID check. by Hung-ying Tyan · 14 years ago
  13. a77c954 am cbee6229: am 0a537b78: Merge "RTP: Enable AMR codec." into gingerbread by Chia-chi Yeh · 14 years ago
  14. d161479 am 947d2abd: am 2365b78e: Merge "SIP: misc fixes." into gingerbread by Hung-ying Tyan · 14 years ago
  15. 5a7c6d2 am 1c2eab2d: am 955ab37c: Merge "RTP: Enable GSM-EFR codec." into gingerbread by Chia-chi Yeh · 14 years ago
  16. 21ae1ad RTP: Minor fixes with polishing. by Chia-chi Yeh · 14 years ago
  17. d29e075 Merge "Add uri field to SipManager.ListenerRelay" into gingerbread by Hung-ying Tyan · 14 years ago
  18. 9e1d308 Add uri field to SipManager.ListenerRelay by Hung-ying Tyan · 14 years ago
  19. dfd1484 Merge "RTP: Adjust the jitter buffer to 512ms." into gingerbread by Chia-chi Yeh · 14 years ago
  20. 3520bd4 RTP: Adjust the jitter buffer to 512ms. by Chia-chi Yeh · 14 years ago
  21. 8ff0722 am 1254b9c5: am cd386649: Merge "RTP: Revise the workaround of private addresses and fix bugs." into gingerbread by Chia-chi Yeh · 14 years ago
  22. 6a53489 SipService: add UID check. by Hung-ying Tyan · 14 years ago
  23. 0a537b7 Merge "RTP: Enable AMR codec." into gingerbread by Chia-chi Yeh · 14 years ago
  24. 2365b78 Merge "SIP: misc fixes." into gingerbread by Hung-ying Tyan · 14 years ago
  25. f88fc1f RTP: Enable AMR codec. by Chia-chi Yeh · 14 years ago
  26. fb3a98b SIP: misc fixes. by Hung-ying Tyan · 14 years ago
  27. f4ae942 RTP: Enable GSM-EFR codec. by Chia-chi Yeh · 14 years ago
  28. fe52989 RTP: Revise the workaround of private addresses and fix bugs. by Chia-chi Yeh · 14 years ago
  29. dcf2be6 am ebfe5632: am e006e4d2: Merge changes Iae1913fb,I38dbefef into gingerbread by Chia-chi Yeh · 14 years ago
  30. e006e4d Merge changes Iae1913fb,I38dbefef into gingerbread by Chia-chi Yeh · 14 years ago
  31. a6f950c RTP: Enable GSM codec. by Chia-chi Yeh · 14 years ago
  32. 9783052 am df31e03c: am 320cdcb1: Merge "RTP: Delay the initialization of AudioTrack and AudioRecord." into gingerbread by Chia-chi Yeh · 14 years ago
  33. 0b3968a am 0d447760: am 6d028dd2: Merge "SIP: Feedback any provisional responses in addition to RING" into gingerbread by Hung-ying Tyan · 14 years ago
  34. 78c11b3 RTP: Refactor out G711 codecs into another file. by Chia-chi Yeh · 14 years ago
  35. 320cdcb Merge "RTP: Delay the initialization of AudioTrack and AudioRecord." into gingerbread by Chia-chi Yeh · 14 years ago
  36. 9083c84 RTP: Delay the initialization of AudioTrack and AudioRecord. by Chia-chi Yeh · 14 years ago
  37. 6c6eacd am f7e13400: am 624d5b4e: SIP: add DisconnectCause.SERVER_ERROR by Hung-ying Tyan · 14 years ago
  38. 6057cd0 SIP: Feedback any provisional responses in addition to RING by Hung-ying Tyan · 14 years ago
  39. 624d5b4 SIP: add DisconnectCause.SERVER_ERROR by Hung-ying Tyan · 14 years ago
  40. a57afb6 resolved conflicts for merge of 2a36a778 to master by Hung-ying Tyan · 14 years ago
  41. 7e54ef7 Move SipService out of SystemServer to phone process. by Hung-ying Tyan · 14 years ago
  42. 5a474a2 am 44669d31: am fd144d76: Merge "SipAudioCall: remove SipManager dependency." into gingerbread by Hung-ying Tyan · 14 years ago
  43. 031d878 am fe2d279c: am 00a22064: SipService: handle cross-domain authentication error by Hung-ying Tyan · 14 years ago
  44. fd144d7 Merge "SipAudioCall: remove SipManager dependency." into gingerbread by Hung-ying Tyan · 14 years ago
  45. 00a2206 SipService: handle cross-domain authentication error by Hung-ying Tyan · 14 years ago
  46. 5e18ad0 am 4a04a312: am bd229420: Fix the unhold issue especially if one is behind NAT. by Chung-yih Wang · 14 years ago
  47. bd22942 Fix the unhold issue especially if one is behind NAT. by Chung-yih Wang · 14 years ago
  48. 3a4197e SipAudioCall: remove SipManager dependency. by Hung-ying Tyan · 14 years ago
  49. a97c5f7 Merge "fix build" by Hung-ying Tyan · 14 years ago
  50. fb02640 fix build by Hung-ying Tyan · 14 years ago
  51. 658bec9 SDP: remove dead code. by Chia-chi Yeh · 14 years ago
  52. 84a357b Refactoring SIP classes to get ready for API review. by Hung-ying Tyan · 14 years ago
  53. 0b7d6de Fix the build. by repo sync · 14 years ago
  54. 84f7f6b SIP: Make SipAudioCallImpl use SimpleSessionDescription instead of javax.sdp. by repo sync · 14 years ago
  55. e6c0c10 SDP: Add a simple class to help manipulate session descriptions. by Chia-chi Yeh · 14 years ago
  56. 7a69aef RTP: Add log throttle for "no data". by repo sync · 14 years ago
  57. 4033a67 RTP: Update native part to reflect the API change. by Chia-chi Yeh · 14 years ago
  58. 37adc52 RTP: Add two getters to retrieve the current configuration from AudioStream. by Chia-chi Yeh · 14 years ago
  59. 32e106b RTP: Extend codec capability and update the APIs. by Chia-chi Yeh · 14 years ago
  60. 8544560 SipPhone: fix missing-call DisconnectCause feedback by Hung-ying Tyan · 14 years ago
  61. 9796379 SIP: convert enum to static final int. by Hung-ying Tyan · 14 years ago
  62. c4b8747 SIP: add config flag for wifi-only configuration. by Hung-ying Tyan · 14 years ago
  63. afa583e SipAudioCall: expose startAudio() by Hung-ying Tyan · 14 years ago
  64. 9352cf1 Add timer to SIP session creation process. by Hung-ying Tyan · 14 years ago
  65. 286bb5a Fix links in SIP API javadoc. by Hung-ying Tyan · 14 years ago
  66. ae076d3 SIP: add PEER_NOT_REACHABLE error feedback. by Hung-ying Tyan · 14 years ago
  67. 12bec5d SipService: ignore connect event for non-active networks. by Hung-ying Tyan · 14 years ago
  68. 13f6270 SipAudioCall: use SipErrorCode instead of string in onError() by Hung-ying Tyan · 14 years ago
  69. 99bf4e4 SIP: remove dependency on javax.sip by Hung-ying Tyan · 14 years ago
  70. d231aa8 SipService: deliver connectivity change to all sessions. by Hung-ying Tyan · 14 years ago
  71. 3d7606a SIP: enhance timeout and registration status feedback. by Hung-ying Tyan · 14 years ago
  72. 25b52a2 SIP: remove dependency on javax.sip.SipException. by Hung-ying Tyan · 14 years ago
  73. 903e103 SIP: add SipErrorCode for error feedback. by Hung-ying Tyan · 14 years ago
  74. f6936a3a Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread by Chia-chi Yeh · 14 years ago
  75. 557b04d RTP: prevent buffer overflow in AudioRecord. by Chia-chi Yeh · 14 years ago
  76. 643fce9 SipManager: always return true for SIP API and VOIP support query. by Hung-ying Tyan · 14 years ago
  77. dc296b0 Merge "SipService: reduce the usage of javax.sdp.*." into gingerbread by Chia-chi Yeh · 14 years ago
  78. 95b15c3 SipService: reduce the usage of javax.sdp.*. by Chia-chi Yeh · 14 years ago
  79. 60264b30 SipProfile: remove outgoingCallAllowed flag. by Hung-ying Tyan · 14 years ago
  80. 3424c02 Add software features for SIP and VOIP by Hung-ying Tyan · 14 years ago
  81. 0858806 Add Wifi High Perf. mode during a call. by Chung-yih Wang · 14 years ago
  82. 14e0062 Merge "Revert "RTP: integrate the echo canceller from speex."" into gingerbread by Chia-chi Yeh · 14 years ago
  83. 7fa7ee1 Revert "RTP: integrate the echo canceller from speex." by Chia-chi Yeh · 14 years ago
  84. 5424c8d Add dynamic uid info for tracking the sip service usage. by Chung-yih Wang · 14 years ago
  85. 37f709a Merge "SipProfile: add isOutgoingCallAllowed() and new builder constructor" into gingerbread by Hung-ying Tyan · 14 years ago
  86. cf95f5d SipProfile: add isOutgoingCallAllowed() and new builder constructor by Hung-ying Tyan · 14 years ago
  87. 3294d44 Add confcall management to SIP calls by Hung-ying Tyan · 14 years ago
  88. 4ae6ec4 RTP: integrate the echo canceller from speex. by Chia-chi Yeh · 14 years ago
  89. 2880ef8 RTP: reduce the latency by overlapping AudioRecord and AudioTrack. by Chia-chi Yeh · 14 years ago
  90. b879032 RTP: fix few leaks when fail to add streams into a group. by Chia-chi Yeh · 14 years ago
  91. 3459d30 RTP: remove froyo-compatible code. by Chia-chi Yeh · 14 years ago
  92. cfd15dd Fix the IN_CALL mode issue. by Chung-yih Wang · 14 years ago
  93. ea4de5bd SipAudioCall: perform local ops before network op in endCall() by Hung-ying Tyan · 14 years ago
  94. 8e63ddb SIP: clean up unused class and fields. by Hung-ying Tyan · 14 years ago
  95. 4c5d28c RTP: move into frameworks. by Chia-chi Yeh · 14 years ago
  96. cde66df Revert "Move SIP telephony related codes to framework." by Chung-yih Wang · 14 years ago
  97. b631dcf Move SIP telephony related codes to framework. by Chung-yih Wang · 14 years ago
  98. 363c2ab Move the sip related codes to framework. by Chung-yih Wang · 14 years ago