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gerrit-public.fairphone.software / platform / frameworks / base / 191713a51f0292b7fce72cc7e16cfd0f0204a0c7 / voip
  1. 3762c31 Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGE by Steve Block · 13 years ago
  2. 8564c8d Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGE by Steve Block · 13 years ago
  3. 6215d3f Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE by Steve Block · 13 years ago
  4. 5baa3a6 Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF) DO NOT MERGE by Steve Block · 13 years ago
  5. c2bd616 SipService: grab Wi-Fi lock only when necessary. by Chia-chi Yeh · 13 years ago
  6. 2b07267 am 201469f5: am bb7f590a: Merge "docs: Add developer guide cross-references, Project ACRE, round 4" into ics-mr1 by Joe Fernandez · 13 years ago
  7. 3aef8e1d docs: Add developer guide cross-references, Project ACRE, round 4 by Joe Fernandez · 13 years ago
  8. 98c54a3 am 0e67685e: am bfd85f9a: Merge "SIP: turn off verbose logs." into ics-mr1 by Chia-chi Yeh · 13 years ago
  9. cb6ee06 SIP: turn off verbose logs. by Chia-chi Yeh · 13 years ago
  10. 71f2cf1 Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE by Steve Block · 13 years ago
  11. ee59e6a SipService: handle connectivity changes correctly. by Chia-chi Yeh · 13 years ago
  12. 6d8b9b8 Merge "RTP: Update parameters for larger packet intervals." by Chia-chi Yeh · 13 years ago
  13. 7a685e8 Merge "SIP: fix keep-alive measurement and increase the timeout." by Chia-chi Yeh · 13 years ago
  14. d17b6d5 SIP: fix keep-alive measurement and increase the timeout. by Chia-chi Yeh · 13 years ago
  15. be57bfe RTP: Update parameters for larger packet intervals. by Chia-chi Yeh · 13 years ago
  16. 81a5ec5 Merge "RTP: support payloads with larger packetization interval." by Chia-chi Yeh · 13 years ago
  17. fa6067f Merge "VoIP JNI: Force AEC on for tuna board" by Eric Laurent · 13 years ago
  18. 35d05dc RTP: support payloads with larger packetization interval. by Chia-chi Yeh · 13 years ago
  19. 54eabd6 SIP: avoid extreme small values in Min-Expires headers. by Chia-chi Yeh · 13 years ago
  20. 74e0a99 VoIP JNI: Force AEC on for tuna board by Eric Laurent · 13 years ago
  21. 5f76006 SIP: add the check for expiry time in Contact header. by Chia-chi Yeh · 13 years ago
  22. dc5bbe9 Handle SIP authentication response for BYE. by Hung-ying Tyan · 13 years ago
  23. 53ad2c7 am 0793586b: am f8c1f129: am e1d27154: am f87743e7: Merge "Prevent NullPointerException cases while using SipService." by Conley Owens · 13 years ago
  24. 0793586 am f8c1f129: am e1d27154: am f87743e7: Merge "Prevent NullPointerException cases while using SipService." by Conley Owens · 13 years ago
  25. 25ccbb9 Prevent NullPointerException cases while using SipService. by Masahiko Endo · 13 years ago
  26. 5fb3ba6 Issue 3370834: No Echo canceler for SIP by Eric Laurent · 13 years ago
  27. 307f15f Add REFER handling. by repo sync · 13 years ago
  28. 3eeb1a9 Merge "Keep last known keepalive interval to avoid duplicate effort." by Hung-ying Tyan · 13 years ago
  29. 9324e04 Merge "Do not hold wifi lock when SIP is also available over mobile network." by Hung-ying Tyan · 13 years ago
  30. f8c34ad Merge "Do not keep alive for re-established call." by Hung-ying Tyan · 13 years ago
  31. 9edfa10 Keep last known keepalive interval to avoid duplicate effort. by Hung-ying Tyan · 13 years ago
  32. 8ba4566 Do not keep alive for re-established call. by Hung-ying Tyan · 13 years ago
  33. f89654d Do not hold wifi lock when SIP is also available over mobile network. by Hung-ying Tyan · 13 years ago
  34. a6cec8f Synchronize SipWakeupTimer.onReceive() by Hung-ying Tyan · 13 years ago
  35. 129d0b0 Make NAT port timeout measurement more flexible. by Hung-ying Tyan · 13 years ago
  36. 99705b5 Record external IP and port from SIP responses by Hung-ying Tyan · 14 years ago
  37. 2093561 Support INVITE w/o SDP. by repo sync · 14 years ago
  38. 233718c Start keepalive process for the caller of a SIP call by Hung-ying Tyan · 14 years ago
  39. 1aceda3 Support Invite w/ Replaces request. by repo sync · 14 years ago
  40. e65f3a8 Restart NAT port timeout measurement when keepalive fails and other fixes by Hung-ying Tyan · 14 years ago
  41. 4af085f Execute all the due wakeup events in SipWakeupTimer. by Hung-ying Tyan · 14 years ago
  42. 1275070 Keep the keepalive process going after NAT port is changed. by Hung-ying Tyan · 14 years ago
  43. 4a267a9 Move the keepalive process to SipSessionImpl and make it reusable. by Hung-ying Tyan · 14 years ago
  44. ac320b2 Merge "Move WakeupTimer out of SipService." by Hung-ying Tyan · 14 years ago
  45. 5621554 Move WakeupTimer out of SipService. by Hung-ying Tyan · 14 years ago
  46. c133781 Fix the issue of onNetwork in UI thread. by repo sync · 14 years ago
  47. bb0a989 Add KeepAlive Interval Measurement. by Chung-yih Wang · 14 years ago
  48. 34bb419 update for new audio.h header location by Dima Zavin · 14 years ago
  49. 4b3913a Squashed commit of the following: by Andreas Huber · 14 years ago
  50. b8df57d am d81214da: am a7a9c4cb: am 46524f83: Merge "docs: add package description for RTP" into honeycomb-mr1 by Scott Main · 14 years ago
  51. d81214d am a7a9c4cb: am 46524f83: Merge "docs: add package description for RTP" into honeycomb-mr1 by Scott Main · 14 years ago
  52. de9acb7 docs: add package description for RTP by Scott Main · 14 years ago
  53. 24fc2fb audio/media: convert to using the audio HAL and new audio defs by Dima Zavin · 14 years ago
  54. d8cbd16 am 7a492a9a: am b7a76e84: am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread by Eric Laurent · 14 years ago
  55. 7a492a9 am b7a76e84: am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread by Eric Laurent · 14 years ago
  56. b7a76e8 am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread by Eric Laurent · 14 years ago
  57. d7a724e Issue 4157048: mic gain for VoIP/SIP calls. by Eric Laurent · 14 years ago
  58. 397de16 am fae5e289: am 6f67e7bf: am 2e383bc6: Merge "Making it possible to call SIP calls with special allowed chars." by Brad Fitzpatrick · 14 years ago
  59. fae5e28 am 6f67e7bf: am 2e383bc6: Merge "Making it possible to call SIP calls with special allowed chars." by Brad Fitzpatrick · 14 years ago
  60. b5c72ea Making it possible to call SIP calls with special allowed chars. by Magnus Strandberg · 14 years ago
  61. 3f9e089 Include strings.h instead of string.h for the strcasecmp prototype. by Carl Shapiro · 14 years ago
  62. 3070af0 frameworks/base: remove LOCAL_PRELINK_MODULE by Iliyan Malchev · 14 years ago
  63. 6defd2d NEW_API: Unhide RTP APIs. by Chia-chi Yeh · 14 years ago
  64. c52f5b2 RTP: update javadocs. by Chia-chi Yeh · 14 years ago
  65. 89bc1fe Activate the wifi high perf. for sip calls. by Chung-yih Wang · 14 years ago
  66. fcd0e50 Add rport argument for a reinvite request. by Chung-yih Wang · 14 years ago
  67. 9e25df4 Make SIP AuthName APIs public. by Chung-yih Wang · 14 years ago
  68. 2ba92c7 do not merge bug 3370834 Cherrypick from master by Jean-Michel Trivi · 14 years ago
  69. 14b6d06 Merge changes Ib70e0cf2,I0691cd70 into gingerbread by Hung-ying Tyan · 14 years ago
  70. f46013b Merge "Merge "SipService: registers broadcast receivers on demand."" into honeycomb by Hung-ying Tyan · 14 years ago
  71. e9b5407 Merge "SipService: registers broadcast receivers on demand." by Hung-ying Tyan · 14 years ago
  72. 40f2cac Merge "SipService: release wake lock for cancelled tasks." by Hung-ying Tyan · 14 years ago
  73. 0f7de88 Merge "Add auth. username in SipProfile." from gingerbread. by Chung-yih Wang · 14 years ago
  74. f268a2f Add auth. username in SipProfile. by Chung-yih Wang · 14 years ago
  75. f0bb1ce SipService: registers broadcast receivers on demand. by Hung-ying Tyan · 14 years ago
  76. d87be27 Enable built-in echo canceler if available. by Chia-chi Yeh · 14 years ago
  77. 4bf82df Do not set back to AudioManager.MODE_NORMAL in SipAudioCall. by Chia-chi Yeh · 14 years ago
  78. 0c01e6e SipService: release wake lock for cancelled tasks. by Hung-ying Tyan · 14 years ago
  79. d0da380 am dc78e3fe: am 3cf71376: RTP: Send silence packets on idle streams for every second. by Chia-chi Yeh · 14 years ago
  80. 3cf7137 RTP: Send silence packets on idle streams for every second. by Chia-chi Yeh · 14 years ago
  81. 33808c6 am aec9a33f: am e0bd2688: Merge "Check if VoIP API is supported in SipManager." into gingerbread by Hung-ying Tyan · 14 years ago
  82. 5bd3782 Check if VoIP API is supported in SipManager. by Hung-ying Tyan · 14 years ago
  83. 635b2b7 am d90bc225: am a936b256: Remove SIP realm/domain check by Hung-ying Tyan · 14 years ago
  84. a936b25 Remove SIP realm/domain check by Hung-ying Tyan · 14 years ago
  85. 58ee2ac Check port in create peer's SIP profile. by Hung-ying Tyan · 14 years ago
  86. eecf4a6 Check port in create peer's SIP profile. by Hung-ying Tyan · 14 years ago
  87. c030a16 am c9cc9ab5: am 5f86d7f5: Merge "Fix SIP bug of different transport/port used for requests." into gingerbread by Chung-yih Wang · 14 years ago
  88. f053292 Fix SIP bug of different transport/port used for requests. by Chung-yih Wang · 14 years ago
  89. 2aef9a1 am 7da1ffc9: am e2abd103: Merge "Set AudioGroup mode according to audio settings" into gingerbread by Hung-ying Tyan · 14 years ago
  90. e2abd10 Merge "Set AudioGroup mode according to audio settings" into gingerbread by Hung-ying Tyan · 14 years ago
  91. d6b0d68 am 6034f9b2: am 06e8cdc0: Fix race between ending and answering a SIP call. by Hung-ying Tyan · 14 years ago
  92. db42452 am ed34b244: am d7116ff1: Merge "Do not suppress error feedback during a SIP call." into gingerbread by Hung-ying Tyan · 14 years ago
  93. 06e8cdc Fix race between ending and answering a SIP call. by Hung-ying Tyan · 14 years ago
  94. 4c7cc83 Merge "RTP: Prepare to unhide the APIs." by Chia-chi Yeh · 14 years ago
  95. 53aa6ef RTP: Prepare to unhide the APIs. by Chia-chi Yeh · 14 years ago
  96. 1c8c173 am c41b27e2: am 349f3509: Merge "Correct SipService.isOpened() implementation." into gingerbread by Hung-ying Tyan · 14 years ago
  97. 1210067 am 5c85338d: am d9e12303: Merge "Notify SipSessions before closing SIP stack." into gingerbread by Hung-ying Tyan · 14 years ago
  98. ebf28fa am 0e58a952: am 0bba9535: Merge "Throw proper exceptions in SipManager" into gingerbread by Hung-ying Tyan · 14 years ago
  99. 342a9be am e843dfa8: am bd399b0b: Merge "RTP: Pause echo suppressor when far-end volume is low." into gingerbread by Chia-chi Yeh · 14 years ago
  100. fa81463 Set AudioGroup mode according to audio settings by Hung-ying Tyan · 14 years ago