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* Merge "SIP: add SERVER_UNREACHABLE error code." into gingerbreadHung-ying Tyan2010-10-042-2/+7
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| * SIP: add SERVER_UNREACHABLE error code.Hung-ying Tyan2010-10-052-2/+7
* | SipService: supply PendingIntent when open a profile.Hung-ying Tyan2010-10-053-64/+74
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* Merge "RTP: Add a baseline echo suppressor." into gingerbreadChia-chi Yeh2010-10-044-2/+228
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| * RTP: Add a baseline echo suppressor.Chia-chi Yeh2010-10-054-2/+228
* | SIP: minor fixes.Hung-ying Tyan2010-10-042-7/+25
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* Merge "RTP: Start AudioRecord before AudioTrack to avoid being disabled." int...Chia-chi Yeh2010-09-301-14/+13
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| * RTP: Start AudioRecord before AudioTrack to avoid being disabled.Chia-chi Yeh2010-10-011-14/+13
* | Merge "SipService: turn off verbose logging" into gingerbreadHung-ying Tyan2010-09-301-1/+2
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| * SipService: turn off verbose loggingHung-ying Tyan2010-10-011-1/+2
* | RTP: Minor fixes with polishing.Chia-chi Yeh2010-09-303-19/+20
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* Merge "Add uri field to SipManager.ListenerRelay" into gingerbreadHung-ying Tyan2010-09-301-10/+12
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| * Add uri field to SipManager.ListenerRelayHung-ying Tyan2010-09-301-10/+12
* | Merge "RTP: Adjust the jitter buffer to 512ms." into gingerbreadChia-chi Yeh2010-09-291-23/+35
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| * RTP: Adjust the jitter buffer to 512ms.Chia-chi Yeh2010-09-301-23/+35
* | SipService: add UID check.Hung-ying Tyan2010-09-301-15/+65
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* Merge "RTP: Enable AMR codec." into gingerbreadChia-chi Yeh2010-09-293-2/+174
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| * RTP: Enable AMR codec.Chia-chi Yeh2010-09-303-2/+174
* | Merge "SIP: misc fixes." into gingerbreadHung-ying Tyan2010-09-293-58/+94
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| * SIP: misc fixes.Hung-ying Tyan2010-09-303-58/+94
* | RTP: Enable GSM-EFR codec.Chia-chi Yeh2010-09-304-3/+110
* | RTP: Revise the workaround of private addresses and fix bugs.Chia-chi Yeh2010-09-301-27/+24
* | Merge changes Iae1913fb,I38dbefef into gingerbreadChia-chi Yeh2010-09-285-123/+224
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| * RTP: Enable GSM codec.Chia-chi Yeh2010-09-294-4/+84
| * RTP: Refactor out G711 codecs into another file.Chia-chi Yeh2010-09-293-120/+141
* | Merge "RTP: Delay the initialization of AudioTrack and AudioRecord." into gin...Chia-chi Yeh2010-09-281-133/+123
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| * | RTP: Delay the initialization of AudioTrack and AudioRecord.Chia-chi Yeh2010-09-291-133/+123
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* | SIP: Feedback any provisional responses in addition to RINGHung-ying Tyan2010-09-291-1/+5
* | SIP: add DisconnectCause.SERVER_ERRORHung-ying Tyan2010-09-281-5/+2
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* Move SipService out of SystemServer to phone process.Hung-ying Tyan2010-09-284-0/+3308
* 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
* | SipService: handle cross-domain authentication errorHung-ying Tyan2010-09-271-0/+5
* | Fix the unhold issue especially if one is behind NAT.Chung-yih Wang2010-09-271-2/+27
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* SDP: remove dead code.Chia-chi Yeh2010-09-243-530/+0
* 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
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