1. 09225a4 am c0f2d952: am 0b7d6de1: Fix the build. by repo sync · 14 years ago
  2. 6b762aa resolved conflicts for merge of 2b652855 to master by Chia-chi Yeh · 14 years ago
  3. 0b7d6de Fix the build. by repo sync · 14 years ago
  4. 84f7f6b SIP: Make SipAudioCallImpl use SimpleSessionDescription instead of javax.sdp. by repo sync · 14 years ago
  5. e6c0c10 SDP: Add a simple class to help manipulate session descriptions. by Chia-chi Yeh · 14 years ago
  6. 7a69aef RTP: Add log throttle for "no data". by repo sync · 14 years ago
  7. 4033a67 RTP: Update native part to reflect the API change. by Chia-chi Yeh · 14 years ago
  8. 37adc52 RTP: Add two getters to retrieve the current configuration from AudioStream. by Chia-chi Yeh · 14 years ago
  9. 32e106b RTP: Extend codec capability and update the APIs. by Chia-chi Yeh · 14 years ago
  10. 5491c7a am 852e5354: am 8544560c: SipPhone: fix missing-call DisconnectCause feedback by Hung-ying Tyan · 14 years ago
  11. 8544560 SipPhone: fix missing-call DisconnectCause feedback by Hung-ying Tyan · 14 years ago
  12. 2417132 am 37d1b249: am 97963794: SIP: convert enum to static final int. by Hung-ying Tyan · 14 years ago
  13. 9796379 SIP: convert enum to static final int. by Hung-ying Tyan · 14 years ago
  14. 03d8a8f resolved conflicts for merge of cf1620d6 to master by Hung-ying Tyan · 14 years ago
  15. c4b8747 SIP: add config flag for wifi-only configuration. by Hung-ying Tyan · 14 years ago
  16. e30fc66 Fix build. by Hung-ying Tyan · 14 years ago
  17. 93f3947 Fix build by Hung-ying Tyan · 14 years ago
  18. 6c62609 resolved conflicts for merge of 394d1e4b to master by Hung-ying Tyan · 14 years ago
  19. afa583e SipAudioCall: expose startAudio() by Hung-ying Tyan · 14 years ago
  20. 6308514 am 156edcc9: am 9404e633: Merge "Add timer to SIP session creation process." into gingerbread by Hung-ying Tyan · 14 years ago
  21. 9352cf1 Add timer to SIP session creation process. by Hung-ying Tyan · 14 years ago
  22. 3cd497c am 6962aff3: am 698ddb0c: Merge "Fix links in SIP API javadoc." into gingerbread by Hung-ying Tyan · 14 years ago
  23. 3e5246b am 8a33e964: am 5306e0a8: Merge "SIP: add PEER_NOT_REACHABLE error feedback." into gingerbread by Hung-ying Tyan · 14 years ago
  24. 286bb5a Fix links in SIP API javadoc. by Hung-ying Tyan · 14 years ago
  25. fbfb6f2 am 188a8f1b: am 89a7180a: Merge "SipService: ignore connect event for non-active networks." into gingerbread by Hung-ying Tyan · 14 years ago
  26. b1c4a01 am ae83faa3: am 13f6270e: SipAudioCall: use SipErrorCode instead of string in onError() by Hung-ying Tyan · 14 years ago
  27. fcc7fb8 am 3692af92: am 99bf4e45: SIP: remove dependency on javax.sip by Hung-ying Tyan · 14 years ago
  28. 074663c am ca83c25d: am 4565933f: Merge "SipService: deliver connectivity change to all sessions." into gingerbread by Hung-ying Tyan · 14 years ago
  29. ae076d3 SIP: add PEER_NOT_REACHABLE error feedback. by Hung-ying Tyan · 14 years ago
  30. 12bec5d SipService: ignore connect event for non-active networks. by Hung-ying Tyan · 14 years ago
  31. 13f6270 SipAudioCall: use SipErrorCode instead of string in onError() by Hung-ying Tyan · 14 years ago
  32. 99bf4e4 SIP: remove dependency on javax.sip by Hung-ying Tyan · 14 years ago
  33. d231aa8 SipService: deliver connectivity change to all sessions. by Hung-ying Tyan · 14 years ago
  34. 0b5a8bd am a5dce0c1: am 3d7606aa: SIP: enhance timeout and registration status feedback. by Hung-ying Tyan · 14 years ago
  35. 3d7606a SIP: enhance timeout and registration status feedback. by Hung-ying Tyan · 14 years ago
  36. e765994 am 38dc67f4: am 25b52a2f: SIP: remove dependency on javax.sip.SipException. by Hung-ying Tyan · 14 years ago
  37. 25b52a2 SIP: remove dependency on javax.sip.SipException. by Hung-ying Tyan · 14 years ago
  38. a97ccc0 am 5f93c39c: am ca3c24db: Merge "SIP: add SipErrorCode for error feedback." into gingerbread by Hung-ying Tyan · 14 years ago
  39. 903e103 SIP: add SipErrorCode for error feedback. by Hung-ying Tyan · 14 years ago
  40. 3384e0f am c2eff4a7: am f6936a3a: Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread by Chia-chi Yeh · 14 years ago
  41. f6936a3a Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread by Chia-chi Yeh · 14 years ago
  42. 557b04d RTP: prevent buffer overflow in AudioRecord. by Chia-chi Yeh · 14 years ago
  43. 3559d1d am ea16e72b: am b355714a: Merge "SipManager: always return true for SIP API and VOIP support query." into gingerbread by Hung-ying Tyan · 14 years ago
  44. 643fce9 SipManager: always return true for SIP API and VOIP support query. by Hung-ying Tyan · 14 years ago
  45. 3d67c56 resolved conflicts for merge of 12eaf9d5 to master by repo sync · 14 years ago
  46. dc296b0 Merge "SipService: reduce the usage of javax.sdp.*." into gingerbread by Chia-chi Yeh · 14 years ago
  47. 95b15c3 SipService: reduce the usage of javax.sdp.*. by Chia-chi Yeh · 14 years ago
  48. e93a492 am 18dfae22: am 60264b30: SipProfile: remove outgoingCallAllowed flag. by Hung-ying Tyan · 14 years ago
  49. 60264b30 SipProfile: remove outgoingCallAllowed flag. by Hung-ying Tyan · 14 years ago
  50. f83d4f1 resolved conflicts for merge of 3e4975a5 to master by Hung-ying Tyan · 14 years ago
  51. 3424c02 Add software features for SIP and VOIP by Hung-ying Tyan · 14 years ago
  52. b6d4723 Revert "Add Wifi High Perf. mode during a call." by Chung-yih Wang · 14 years ago
  53. 0858806 Add Wifi High Perf. mode during a call. by Chung-yih Wang · 14 years ago
  54. 14e0062 Merge "Revert "RTP: integrate the echo canceller from speex."" into gingerbread by Chia-chi Yeh · 14 years ago
  55. 7fa7ee1 Revert "RTP: integrate the echo canceller from speex." by Chia-chi Yeh · 14 years ago
  56. 5424c8d Add dynamic uid info for tracking the sip service usage. by Chung-yih Wang · 14 years ago
  57. 37f709a Merge "SipProfile: add isOutgoingCallAllowed() and new builder constructor" into gingerbread by Hung-ying Tyan · 14 years ago
  58. cf95f5d SipProfile: add isOutgoingCallAllowed() and new builder constructor by Hung-ying Tyan · 14 years ago
  59. 3294d44 Add confcall management to SIP calls by Hung-ying Tyan · 14 years ago
  60. 4ae6ec4 RTP: integrate the echo canceller from speex. by Chia-chi Yeh · 14 years ago
  61. 2880ef8 RTP: reduce the latency by overlapping AudioRecord and AudioTrack. by Chia-chi Yeh · 14 years ago
  62. b879032 RTP: fix few leaks when fail to add streams into a group. by Chia-chi Yeh · 14 years ago
  63. 3459d30 RTP: remove froyo-compatible code. by Chia-chi Yeh · 14 years ago
  64. cfd15dd Fix the IN_CALL mode issue. by Chung-yih Wang · 14 years ago
  65. ea4de5bd SipAudioCall: perform local ops before network op in endCall() by Hung-ying Tyan · 14 years ago
  66. 8e63ddb SIP: clean up unused class and fields. by Hung-ying Tyan · 14 years ago
  67. 4c5d28c RTP: move into frameworks. by Chia-chi Yeh · 14 years ago
  68. cde66df Revert "Move SIP telephony related codes to framework." by Chung-yih Wang · 14 years ago
  69. b631dcf Move SIP telephony related codes to framework. by Chung-yih Wang · 14 years ago
  70. 363c2ab Move the sip related codes to framework. by Chung-yih Wang · 14 years ago