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AgeCommit message (Expand)Author
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
2010-07-06SipAudioCallImpl: revert the changes to hold/unhold implementation.Hung-ying Tyan
2010-07-02SIP telephony: single call works (both incoming and outgoing).Hung-ying Tyan
2010-07-02SipAudioCall: re-implemented holding/unholding a call.Hung-ying Tyan
2010-07-02SIP: add call busy handling to demo in-call screenHung-ying Tyan
2010-07-02SipAudioCall: add new setListener() to explicitly specify immediate callbackHung-ying Tyan
2010-07-01SIP: fix two bugs.Hung-ying Tyan
2010-06-30SIP telephony: add receiving call support (roughly)Hung-ying Tyan
2010-06-30SIP telephony: work-in-progresHung-ying Tyan
2010-06-29SIP telephony: mv SipPhoneFactory to where it should be.Hung-ying Tyan
2010-06-28SIP: work-in-progress for telephony integration.Hung-ying Tyan
2010-06-25SIP: duplicate PhoneApp for telephony integration developmentHung-ying Tyan
2010-06-25ISipService: add new open(), open3(), getListOfProfiles()Hung-ying Tyan
2010-06-23SIP: change copyright yearHung-ying Tyan
2010-06-23SIP: duplicate PhoneApp for telephony integration developmentHung-ying Tyan
2010-06-23SIP: first check-in of SipPhone and related classes.Hung-ying Tyan
2010-06-23SIP: duplicate PhoneApp for telephony integration developmentHung-ying Tyan
2010-06-23SIP: add sendDtmf() with callbackHung-ying Tyan
2010-06-24Add the missing resource file.Chung-yih Wang
2010-06-23SIP: rearrange src files to separate settings and demo from framework codeHung-ying Tyan
2010-06-18Some sip setting changes and registration fix.Chung-yih Wang
2010-06-11SipCallUi: enable speaker and end-call buttons when making callHung-ying Tyan
2010-06-11SIP: SipAudioCallImpl: make ringback tone STREAM_VOICE_CALLHung-ying Tyan
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