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* | SipSessionImpl: add MinExpiresHeader check.Hung-ying Tyan2010-07-231-1/+6
* | SipSessionImpl: don't end call when an error occurs during a call.Hung-ying Tyan2010-07-231-65/+41
* | Merge "SipAudioCallImpl: deliver call change failure and don't end call when ...Hung-ying Tyan2010-07-221-4/+13
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| * | SipAudioCallImpl: deliver call change failure and don't end call when getting...Hung-ying Tyan2010-07-221-4/+13
* | | Merge "SIP telephony: don't end the call when getting error in a call."Hung-ying Tyan2010-07-221-10/+15
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| * | SIP telephony: don't end the call when getting error in a call.Hung-ying Tyan2010-07-221-10/+15
* | | Merge "SIP: demo call UI: hold call in onPause() and unhold in onResume() to ...Hung-ying Tyan2010-07-221-4/+11
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| * | SIP: demo call UI: hold call in onPause() and unhold in onResume() to make it...Hung-ying Tyan2010-07-221-4/+11
* | | Use SIP OPTIONS instead of EMPTY message for keep-alive.Chung-yih Wang2010-07-236-38/+125
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* | Merge changes I9adc67d2,I32dd22afChung-yih Wang2010-07-227-125/+229
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| * | SIP: fix a recursion bug when local IP becomes invalid (network disconnected).Hung-ying Tyan2010-07-211-1/+1
| * | SIP telephony: integrate with new RTP stack and other fixes.Hung-ying Tyan2010-07-216-124/+228
* | | RTP: use safer frame count to create AudioTrack and AudioRecord.Chia-chi Yeh2010-07-201-5/+2
* | | RTP: drain DeviceSocket before starting DeviceThread.Chia-chi Yeh2010-07-201-1/+5
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* | SIP telephony: add holding/swapping-calls, call-waitingHung-ying Tyan2010-07-143-28/+154
* | RTP: temporarily make it froyo compatible to ease the development.Chia-chi Yeh2010-07-141-0/+75
* | RTP: tweak the lower bound of buffer size.Chia-chi Yeh2010-07-141-1/+8
* | RTP: add missing string.h to RtpStream.cpp.Chia-chi Yeh2010-07-121-0/+1
* | RTP: remove trailing spaces and add few logs.Chia-chi Yeh2010-07-122-21/+23
* | RTP: add the missing file for librtp_jni.Chia-chi Yeh2010-07-071-0/+32
* | RTP: add Java AudioGroup.Chia-chi Yeh2010-07-071-0/+90
* | RTP: move AudioCodec to android.net.rtp.Chia-chi Yeh2010-07-071-0/+34
* | RTP: refactor out the network part from AudioStream to RtpStream.Chia-chi Yeh2010-07-072-0/+307
* | RTP: add glue code for jni part.Chia-chi Yeh2010-07-071-0/+44
* | RTP: add AudioGroup which handles conference call, jitter buffer, and more.Chia-chi Yeh2010-07-071-0/+919
* | RTP: abstract the network part from AudioStream to RtpStream.Chia-chi Yeh2010-07-072-0/+186
* | RTP: refactor native audio codec.Chia-chi Yeh2010-07-072-0/+197
* | SIP: cross out password when a profile is added to SipServiceHung-ying Tyan2010-07-074-45/+94
* | SipAudioCallImpl: revert the changes to hold/unhold implementation.Hung-ying Tyan2010-07-061-16/+0
* | SIP telephony: single call works (both incoming and outgoing).Hung-ying Tyan2010-07-027-600/+186
* | SipAudioCall: re-implemented holding/unholding a call.Hung-ying Tyan2010-07-021-0/+16
* | SIP: add call busy handling to demo in-call screenHung-ying Tyan2010-07-021-0/+6
* | SipAudioCall: add new setListener() to explicitly specify immediate callbackHung-ying Tyan2010-07-022-27/+56
* | SIP: fix two bugs.Hung-ying Tyan2010-07-012-5/+16
* | SIP telephony: add receiving call support (roughly)Hung-ying Tyan2010-06-306-9/+76
* | SIP telephony: work-in-progresHung-ying Tyan2010-06-304-118/+284
* | SIP telephony: mv SipPhoneFactory to where it should be.Hung-ying Tyan2010-06-2931-130/+124
* | SIP: work-in-progress for telephony integration.Hung-ying Tyan2010-06-2894-2759/+4207
* | SIP: duplicate PhoneApp for telephony integration developmentHung-ying Tyan2010-06-2528-900/+516
* | ISipService: add new open(), open3(), getListOfProfiles()Hung-ying Tyan2010-06-256-32/+183
* | SIP: change copyright yearHung-ying Tyan2010-06-235-5/+5
* | SIP: duplicate PhoneApp for telephony integration developmentHung-ying Tyan2010-06-2375-1232/+1306
* | SIP: first check-in of SipPhone and related classes.Hung-ying Tyan2010-06-235-0/+1991
* | SIP: duplicate PhoneApp for telephony integration developmentHung-ying Tyan2010-06-23500-0/+54788
* | SIP: add sendDtmf() with callbackHung-ying Tyan2010-06-232-3/+18
* | Add the missing resource file.Chung-yih Wang2010-06-243-0/+60
* | SIP: rearrange src files to separate settings and demo from framework codeHung-ying Tyan2010-06-2312-0/+41
* | Some sip setting changes and registration fix.Chung-yih Wang2010-06-187-86/+140
* | SipCallUi: enable speaker and end-call buttons when making callHung-ying Tyan2010-06-111-15/+24
* | SIP: SipAudioCallImpl: make ringback tone STREAM_VOICE_CALLHung-ying Tyan2010-06-111-2/+10
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