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2010-08-06RTP: remove dead code. Now we are officially in the frameworks.Chia-chi Yeh
2010-08-06RTP: prepare for moving into frameworks.Chia-chi Yeh
2010-08-05am c05783e9: Merge changes I45517c7b,Ib13825c9 into gingerbreadChung-yih Wang
2010-08-05Merge changes I45517c7b,Ib13825c9 into gingerbreadChung-yih Wang
2010-08-06Fix the build.Chung-yih Wang
2010-08-05am 4b560d4a: Cherrypick the change for setRPort() in Via header.Chung-yih Wang
2010-08-05Cherrypick the change for setRPort() in Via header.Chung-yih Wang
2010-08-05Cherry-pick the change from master for the setRPort in Via header.Chung-yih Wang
2010-08-04Merge "SIP: the low volume problem when setting IN_CALL mode."Alan Huang
2010-08-05SIP: the low volume problem when setting IN_CALL mode.Alan Huang
2010-08-03Generated hdpi drawable size reducedGilles Debunne
2010-08-03external/nist-sip/ : added scaled hdpi version of assets that were moved to d...Gilles Debunne
2010-08-02Move pngs from drawable to drawable-mdpi in project external/nist-sip/Gilles Debunne
2010-07-28Add the missing functions for compatibility.Chung-yih Wang
2010-07-28Fix autoregistration bug.Chung-yih Wang
2010-07-27SIP: demo app: pass on phone number when OutgoingCallReceiver cannot handle it.Hung-ying Tyan
2010-07-26SIP telephony: fix call log and connection stats.Hung-ying Tyan
2010-07-23SIP: fix WakeupTimer.recalculatePeriods().Hung-ying Tyan
2010-07-23SipService: fix NAT binding lifetime measurement.Hung-ying Tyan
2010-07-23SIP: fix a bug in WakeupTimer.cancel()Hung-ying Tyan
2010-07-23SIP telephony: clean up TODO's. B)Hung-ying Tyan
2010-07-23SipSessionGroup: generating 32-bit random number for tag.Hung-ying Tyan
2010-07-23SipSessionImpl: add MinExpiresHeader check.Hung-ying Tyan
2010-07-23SipSessionImpl: don't end call when an error occurs during a call.Hung-ying Tyan
2010-07-22Merge "SipAudioCallImpl: deliver call change failure and don't end call when ...Hung-ying Tyan
2010-07-22Merge "SIP telephony: don't end the call when getting error in a call."Hung-ying Tyan
2010-07-22Merge "SIP: demo call UI: hold call in onPause() and unhold in onResume() to ...Hung-ying Tyan
2010-07-23 Use SIP OPTIONS instead of EMPTY message for keep-alive.Chung-yih Wang
2010-07-22SipAudioCallImpl: deliver call change failure and don't end call when getting...Hung-ying Tyan
2010-07-22SIP telephony: don't end the call when getting error in a call.Hung-ying Tyan
2010-07-22SIP: demo call UI: hold call in onPause() and unhold in onResume() to make it...Hung-ying Tyan
2010-07-22Merge changes I9adc67d2,I32dd22afChung-yih Wang
2010-07-21SIP: fix a recursion bug when local IP becomes invalid (network disconnected).Hung-ying Tyan
2010-07-21SIP telephony: integrate with new RTP stack and other fixes.Hung-ying Tyan
2010-07-20RTP: use safer frame count to create AudioTrack and AudioRecord.Chia-chi Yeh
2010-07-20RTP: drain DeviceSocket before starting DeviceThread.Chia-chi Yeh
2010-07-14SIP telephony: add holding/swapping-calls, call-waitingHung-ying Tyan
2010-07-14RTP: temporarily make it froyo compatible to ease the development.Chia-chi Yeh
2010-07-14RTP: tweak the lower bound of buffer size.Chia-chi Yeh
2010-07-12RTP: add missing string.h to RtpStream.cpp.Chia-chi Yeh
2010-07-12RTP: remove trailing spaces and add few logs.Chia-chi Yeh
2010-07-07RTP: add the missing file for librtp_jni.Chia-chi Yeh
2010-07-07RTP: add Java AudioGroup.Chia-chi Yeh
2010-07-07RTP: move AudioCodec to android.net.rtp.Chia-chi Yeh
2010-07-07RTP: refactor out the network part from AudioStream to RtpStream.Chia-chi Yeh
2010-07-07RTP: add glue code for jni part.Chia-chi Yeh
2010-07-07RTP: add AudioGroup which handles conference call, jitter buffer, and more.Chia-chi Yeh
2010-07-07RTP: abstract the network part from AudioStream to RtpStream.Chia-chi Yeh
2010-07-07RTP: refactor native audio codec.Chia-chi Yeh
2010-07-07SIP: cross out password when a profile is added to SipServiceHung-ying Tyan
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