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* resolved conflicts for merge of 2a36a778 to masterHung-ying Tyan2010-09-284-0/+3308
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| * Move SipService out of SystemServer to phone process.Hung-ying Tyan2010-09-284-0/+3308
* | am 44669d31: am fd144d76: Merge "SipAudioCall: remove SipManager dependency."...Hung-ying Tyan2010-09-272-11/+12
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| * Merge "SipAudioCall: remove SipManager dependency." into gingerbreadHung-ying Tyan2010-09-272-11/+12
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| | * SipAudioCall: remove SipManager dependency.Hung-ying Tyan2010-09-242-11/+12
* | | am fe2d279c: am 00a22064: SipService: handle cross-domain authentication errorHung-ying Tyan2010-09-271-0/+5
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| * | SipService: handle cross-domain authentication errorHung-ying Tyan2010-09-271-0/+5
* | | am 4a04a312: am bd229420: Fix the unhold issue especially if one is behind NAT.Chung-yih Wang2010-09-271-2/+27
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| * | Fix the unhold issue especially if one is behind NAT.Chung-yih Wang2010-09-271-2/+27
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* | Merge "fix build"Hung-ying Tyan2010-09-231-0/+2
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| * fix buildHung-ying Tyan2010-09-241-0/+2
* | SDP: remove dead code.Chia-chi Yeh2010-09-243-530/+0
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* Refactoring SIP classes to get ready for API review.Hung-ying Tyan2010-09-246-1061/+1459
* Fix the build.repo sync2010-09-231-1/+1
* SIP: Make SipAudioCallImpl use SimpleSessionDescription instead of javax.sdp.repo sync2010-09-231-196/+185
* SDP: Add a simple class to help manipulate session descriptions.Chia-chi Yeh2010-09-231-0/+612
* RTP: Add log throttle for "no data".repo sync2010-09-231-1/+5
* RTP: Update native part to reflect the API change.Chia-chi Yeh2010-09-233-39/+48
* RTP: Add two getters to retrieve the current configuration from AudioStream.Chia-chi Yeh2010-09-231-1/+20
* RTP: Extend codec capability and update the APIs.Chia-chi Yeh2010-09-234-62/+244
* SipPhone: fix missing-call DisconnectCause feedbackHung-ying Tyan2010-09-201-8/+8
* SIP: convert enum to static final int.Hung-ying Tyan2010-09-209-71/+127
* SIP: add config flag for wifi-only configuration.Hung-ying Tyan2010-09-201-0/+8
* SipAudioCall: expose startAudio()Hung-ying Tyan2010-09-172-61/+72
* Add timer to SIP session creation process.Hung-ying Tyan2010-09-174-27/+81
* Fix links in SIP API javadoc.Hung-ying Tyan2010-09-167-45/+40
* SIP: add PEER_NOT_REACHABLE error feedback.Hung-ying Tyan2010-09-151-0/+3
* SipService: ignore connect event for non-active networks.Hung-ying Tyan2010-09-151-11/+12
* SipAudioCall: use SipErrorCode instead of string in onError()Hung-ying Tyan2010-09-142-24/+52
* SIP: remove dependency on javax.sipHung-ying Tyan2010-09-144-12/+16
* SipService: deliver connectivity change to all sessions.Hung-ying Tyan2010-09-141-1/+4
* SIP: enhance timeout and registration status feedback.Hung-ying Tyan2010-09-131-1/+7
* SIP: remove dependency on javax.sip.SipException.Hung-ying Tyan2010-09-134-7/+40
* SIP: add SipErrorCode for error feedback.Hung-ying Tyan2010-09-106-25/+68
* Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbreadChia-chi Yeh2010-09-091-2/+2
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| * RTP: prevent buffer overflow in AudioRecord.Chia-chi Yeh2010-09-081-2/+2
* | SipManager: always return true for SIP API and VOIP support query.Hung-ying Tyan2010-09-031-0/+6
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* Merge "SipService: reduce the usage of javax.sdp.*." into gingerbreadChia-chi Yeh2010-09-027-38/+27
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| * SipService: reduce the usage of javax.sdp.*.Chia-chi Yeh2010-09-027-38/+27
* | SipProfile: remove outgoingCallAllowed flag.Hung-ying Tyan2010-09-021-23/+1
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* Add software features for SIP and VOIPHung-ying Tyan2010-09-021-9/+24
* Add Wifi High Perf. mode during a call.Chung-yih Wang2010-08-261-1/+28
* Merge "Revert "RTP: integrate the echo canceller from speex."" into gingerbreadChia-chi Yeh2010-08-252-18/+6
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| * Revert "RTP: integrate the echo canceller from speex."Chia-chi Yeh2010-08-262-18/+6
* | Add dynamic uid info for tracking the sip service usage.Chung-yih Wang2010-08-261-0/+19
* | Merge "SipProfile: add isOutgoingCallAllowed() and new builder constructor" i...Hung-ying Tyan2010-08-241-1/+40
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| * | SipProfile: add isOutgoingCallAllowed() and new builder constructorHung-ying Tyan2010-08-241-1/+40
* | | Add confcall management to SIP callsHung-ying Tyan2010-08-242-18/+45
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* | RTP: integrate the echo canceller from speex.Chia-chi Yeh2010-08-242-6/+18
* | RTP: reduce the latency by overlapping AudioRecord and AudioTrack.Chia-chi Yeh2010-08-241-28/+53
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