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gerrit-public.fairphone.software / platform / frameworks / base / f6a04aeeed4bdb89c5ef8db86ece2617b99aca87 / voip
  1. b1c4a01 am ae83faa3: am 13f6270e: SipAudioCall: use SipErrorCode instead of string in onError() by Hung-ying Tyan · 14 years ago
  2. fcc7fb8 am 3692af92: am 99bf4e45: SIP: remove dependency on javax.sip by Hung-ying Tyan · 14 years ago
  3. 074663c am ca83c25d: am 4565933f: Merge "SipService: deliver connectivity change to all sessions." into gingerbread by Hung-ying Tyan · 14 years ago
  4. 13f6270 SipAudioCall: use SipErrorCode instead of string in onError() by Hung-ying Tyan · 14 years ago
  5. 99bf4e4 SIP: remove dependency on javax.sip by Hung-ying Tyan · 14 years ago
  6. d231aa8 SipService: deliver connectivity change to all sessions. by Hung-ying Tyan · 14 years ago
  7. 0b5a8bd am a5dce0c1: am 3d7606aa: SIP: enhance timeout and registration status feedback. by Hung-ying Tyan · 14 years ago
  8. 3d7606a SIP: enhance timeout and registration status feedback. by Hung-ying Tyan · 14 years ago
  9. e765994 am 38dc67f4: am 25b52a2f: SIP: remove dependency on javax.sip.SipException. by Hung-ying Tyan · 14 years ago
  10. 25b52a2 SIP: remove dependency on javax.sip.SipException. by Hung-ying Tyan · 14 years ago
  11. a97ccc0 am 5f93c39c: am ca3c24db: Merge "SIP: add SipErrorCode for error feedback." into gingerbread by Hung-ying Tyan · 14 years ago
  12. 903e103 SIP: add SipErrorCode for error feedback. by Hung-ying Tyan · 14 years ago
  13. 3384e0f am c2eff4a7: am f6936a3a: Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread by Chia-chi Yeh · 14 years ago
  14. f6936a3a Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread by Chia-chi Yeh · 14 years ago
  15. 557b04d RTP: prevent buffer overflow in AudioRecord. by Chia-chi Yeh · 14 years ago
  16. 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
  17. 643fce9 SipManager: always return true for SIP API and VOIP support query. by Hung-ying Tyan · 14 years ago
  18. 3d67c56 resolved conflicts for merge of 12eaf9d5 to master by repo sync · 14 years ago
  19. dc296b0 Merge "SipService: reduce the usage of javax.sdp.*." into gingerbread by Chia-chi Yeh · 14 years ago
  20. 95b15c3 SipService: reduce the usage of javax.sdp.*. by Chia-chi Yeh · 14 years ago
  21. e93a492 am 18dfae22: am 60264b30: SipProfile: remove outgoingCallAllowed flag. by Hung-ying Tyan · 14 years ago
  22. 60264b30 SipProfile: remove outgoingCallAllowed flag. by Hung-ying Tyan · 14 years ago
  23. f83d4f1 resolved conflicts for merge of 3e4975a5 to master by Hung-ying Tyan · 14 years ago
  24. 3424c02 Add software features for SIP and VOIP by Hung-ying Tyan · 14 years ago
  25. b6d4723 Revert "Add Wifi High Perf. mode during a call." by Chung-yih Wang · 14 years ago
  26. 0858806 Add Wifi High Perf. mode during a call. by Chung-yih Wang · 14 years ago
  27. 14e0062 Merge "Revert "RTP: integrate the echo canceller from speex."" into gingerbread by Chia-chi Yeh · 14 years ago
  28. 7fa7ee1 Revert "RTP: integrate the echo canceller from speex." by Chia-chi Yeh · 14 years ago
  29. 5424c8d Add dynamic uid info for tracking the sip service usage. by Chung-yih Wang · 14 years ago
  30. 37f709a Merge "SipProfile: add isOutgoingCallAllowed() and new builder constructor" into gingerbread by Hung-ying Tyan · 14 years ago
  31. cf95f5d SipProfile: add isOutgoingCallAllowed() and new builder constructor by Hung-ying Tyan · 14 years ago
  32. 3294d44 Add confcall management to SIP calls by Hung-ying Tyan · 14 years ago
  33. 4ae6ec4 RTP: integrate the echo canceller from speex. by Chia-chi Yeh · 14 years ago
  34. 2880ef8 RTP: reduce the latency by overlapping AudioRecord and AudioTrack. by Chia-chi Yeh · 14 years ago
  35. b879032 RTP: fix few leaks when fail to add streams into a group. by Chia-chi Yeh · 14 years ago
  36. 3459d30 RTP: remove froyo-compatible code. by Chia-chi Yeh · 14 years ago
  37. cfd15dd Fix the IN_CALL mode issue. by Chung-yih Wang · 14 years ago
  38. ea4de5bd SipAudioCall: perform local ops before network op in endCall() by Hung-ying Tyan · 14 years ago
  39. 8e63ddb SIP: clean up unused class and fields. by Hung-ying Tyan · 14 years ago
  40. 4c5d28c RTP: move into frameworks. by Chia-chi Yeh · 14 years ago
  41. cde66df Revert "Move SIP telephony related codes to framework." by Chung-yih Wang · 14 years ago
  42. b631dcf Move SIP telephony related codes to framework. by Chung-yih Wang · 14 years ago
  43. 363c2ab Move the sip related codes to framework. by Chung-yih Wang · 14 years ago