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gerrit-public.fairphone.software / platform / frameworks / base / ca3f968c6f0acbf94988d00bf21a82a2e8f49b86 / voip
  1. 658bec9 SDP: remove dead code. by Chia-chi Yeh · 14 years ago
  2. 84a357b Refactoring SIP classes to get ready for API review. by Hung-ying Tyan · 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. 8544560 SipPhone: fix missing-call DisconnectCause feedback by Hung-ying Tyan · 14 years ago
  11. 9796379 SIP: convert enum to static final int. by Hung-ying Tyan · 14 years ago
  12. c4b8747 SIP: add config flag for wifi-only configuration. by Hung-ying Tyan · 14 years ago
  13. afa583e SipAudioCall: expose startAudio() by Hung-ying Tyan · 14 years ago
  14. 9352cf1 Add timer to SIP session creation process. by Hung-ying Tyan · 14 years ago
  15. 286bb5a Fix links in SIP API javadoc. by Hung-ying Tyan · 14 years ago
  16. ae076d3 SIP: add PEER_NOT_REACHABLE error feedback. by Hung-ying Tyan · 14 years ago
  17. 12bec5d SipService: ignore connect event for non-active networks. by Hung-ying Tyan · 14 years ago
  18. 13f6270 SipAudioCall: use SipErrorCode instead of string in onError() by Hung-ying Tyan · 14 years ago
  19. 99bf4e4 SIP: remove dependency on javax.sip by Hung-ying Tyan · 14 years ago
  20. d231aa8 SipService: deliver connectivity change to all sessions. by Hung-ying Tyan · 14 years ago
  21. 3d7606a SIP: enhance timeout and registration status feedback. by Hung-ying Tyan · 14 years ago
  22. 25b52a2 SIP: remove dependency on javax.sip.SipException. by Hung-ying Tyan · 14 years ago
  23. 903e103 SIP: add SipErrorCode for error feedback. by Hung-ying Tyan · 14 years ago
  24. f6936a3a Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread by Chia-chi Yeh · 14 years ago
  25. 557b04d RTP: prevent buffer overflow in AudioRecord. by Chia-chi Yeh · 14 years ago
  26. 643fce9 SipManager: always return true for SIP API and VOIP support query. by Hung-ying Tyan · 14 years ago
  27. dc296b0 Merge "SipService: reduce the usage of javax.sdp.*." into gingerbread by Chia-chi Yeh · 14 years ago
  28. 95b15c3 SipService: reduce the usage of javax.sdp.*. by Chia-chi Yeh · 14 years ago
  29. 60264b30 SipProfile: remove outgoingCallAllowed flag. by Hung-ying Tyan · 14 years ago
  30. 3424c02 Add software features for SIP and VOIP by Hung-ying Tyan · 14 years ago
  31. 0858806 Add Wifi High Perf. mode during a call. by Chung-yih Wang · 14 years ago
  32. 14e0062 Merge "Revert "RTP: integrate the echo canceller from speex."" into gingerbread by Chia-chi Yeh · 14 years ago
  33. 7fa7ee1 Revert "RTP: integrate the echo canceller from speex." by Chia-chi Yeh · 14 years ago
  34. 5424c8d Add dynamic uid info for tracking the sip service usage. by Chung-yih Wang · 14 years ago
  35. 37f709a Merge "SipProfile: add isOutgoingCallAllowed() and new builder constructor" into gingerbread by Hung-ying Tyan · 14 years ago
  36. cf95f5d SipProfile: add isOutgoingCallAllowed() and new builder constructor by Hung-ying Tyan · 14 years ago
  37. 3294d44 Add confcall management to SIP calls by Hung-ying Tyan · 14 years ago
  38. 4ae6ec4 RTP: integrate the echo canceller from speex. by Chia-chi Yeh · 14 years ago
  39. 2880ef8 RTP: reduce the latency by overlapping AudioRecord and AudioTrack. by Chia-chi Yeh · 14 years ago
  40. b879032 RTP: fix few leaks when fail to add streams into a group. by Chia-chi Yeh · 14 years ago
  41. 3459d30 RTP: remove froyo-compatible code. by Chia-chi Yeh · 14 years ago
  42. cfd15dd Fix the IN_CALL mode issue. by Chung-yih Wang · 14 years ago
  43. ea4de5bd SipAudioCall: perform local ops before network op in endCall() by Hung-ying Tyan · 14 years ago
  44. 8e63ddb SIP: clean up unused class and fields. by Hung-ying Tyan · 14 years ago
  45. 4c5d28c RTP: move into frameworks. by Chia-chi Yeh · 14 years ago
  46. cde66df Revert "Move SIP telephony related codes to framework." by Chung-yih Wang · 14 years ago
  47. b631dcf Move SIP telephony related codes to framework. by Chung-yih Wang · 14 years ago
  48. 363c2ab Move the sip related codes to framework. by Chung-yih Wang · 14 years ago