Hung-ying Tyan
8544560ccc
SipPhone: fix missing-call DisconnectCause feedback
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also fix delivering bad news before closing a SipAudioCallImpl object so that
apps can get the current audio-call object state before it's closed:
http://b/issue?id=3009262
Change-Id: I94c19dae8b4f252de869e614ec462b19b4ff2077
2010-09-20 13:06:30 +08:00
Hung-ying Tyan
97963794af
SIP: convert enum to static final int.
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Converts SipErrorCode and SipSessionState.
Change-Id: Iee3a465649ea89d395b2336bbd673c25113e5f93
2010-09-20 09:51:31 +08:00
Hung-ying Tyan
c4b87477c0
SIP: add config flag for wifi-only configuration.
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http://b/issue?id=2994029
Change-Id: I328da9b0f8b70d660dbcefffdac8250341792101
2010-09-20 08:03:20 +08:00
Hung-ying Tyan
afa583e655
SipAudioCall: expose startAudio()
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so that apps can start audio when time is right.
Change-Id: I7ae96689d3a8006b34097533bc2434bc3814b82a
2010-09-17 15:58:18 +08:00
Hung-ying Tyan
9352cf1a4d
Add timer to SIP session creation process.
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+ add timer parameter to ISipSession.make/changeCall(),
+ add timer paramter to SipAudioCall.make/answer/hold/continueCall()'s,
+ add timer parameter to SipManager.makeAudioCall(),
+ modify implementation in SipSessionGroup, SipAudioCallImpl accordingly,
+ make SipPhone to use it with 8-second timeout.
http://b/issue?id=2994748
Change-Id: I661a887e5810087ddc5e2318335e2fa427f80ec6
2010-09-17 13:31:05 +08:00
Hung-ying Tyan
286bb5a00b
Fix links in SIP API javadoc.
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Change-Id: I839280fe18502bb576f6e9c9a7948077c02fa570
2010-09-16 03:52:10 +08:00
Hung-ying Tyan
ae076d3981
SIP: add PEER_NOT_REACHABLE error feedback.
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http://b/issue?id=3002033
Change-Id: Ib64b08919d214acbab89945ac19dc113a68e62ad
2010-09-15 11:30:45 +08:00
Hung-ying Tyan
12bec5ddf5
SipService: ignore connect event for non-active networks.
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+ sanity check and remove redundant code.
Change-Id: I4d3e226851ad7fc4d88ddcd0a5c58f7e33b6c14a
2010-09-15 00:49:02 +08:00
Hung-ying Tyan
13f6270eb1
SipAudioCall: use SipErrorCode instead of string in onError()
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and fix callback in setListener().
Change-Id: Ic2622df992a2ad45cb1e3f71736f320897ae8fb3
2010-09-14 21:36:10 +08:00
Hung-ying Tyan
99bf4e45c4
SIP: remove dependency on javax.sip
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and change errorCodeString to errorCode in
SipRegistrationListener.onRegistrationFailed().
Change-Id: Id9618f5a4b0effaed04f8b0dc60347499d9e4501
2010-09-14 20:29:02 +08:00
Hung-ying Tyan
d231aa880a
SipService: deliver connectivity change to all sessions.
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+ add DATA_CONNECTION_LOST to SipErrorCode
+ convert it to Connection.DisconnectCause.LOST_SIGNAL in SipPhone
http://b/issue?id=2992548
Change-Id: Ie8983c1b81077b21f46304cf60b8e61df1ffd241
2010-09-14 08:00:09 +08:00
Hung-ying Tyan
3d7606aa60
SIP: enhance timeout and registration status feedback.
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http://b/issue?id=2984419
http://b/issue?id=2991065
Change-Id: I2d3b1dd3a70079ff347f7256f4684aea07847f4e
2010-09-13 17:45:39 +08:00
Hung-ying Tyan
25b52a2f97
SIP: remove dependency on javax.sip.SipException.
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Change-Id: I77d289bef1b5e7f1ec0c0408d0bbf96c21085cd7
2010-09-13 16:50:12 +08:00
Hung-ying Tyan
903e103160
SIP: add SipErrorCode for error feedback.
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Change-Id: I8b071d4933479b780a403d0bfa30511f4c23ca8f
2010-09-10 17:15:06 +08:00
Chia-chi Yeh
f6936a3a52
Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread
2010-09-09 02:37:07 -07:00
Chia-chi Yeh
557b04de23
RTP: prevent buffer overflow in AudioRecord.
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This change simply reduces the receive timeout of DeviceSocket. It works
because AudioRecord will block us till there is enough data, which makes
AudioSocket overlap AudioRecord.
Change-Id: I4700224fb407e148ef359a9d99279e10240128d0
2010-09-08 09:56:02 +08:00
Hung-ying Tyan
643fce9781
SipManager: always return true for SIP API and VOIP support query.
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Change-Id: I397a804e0aa598aee77a8ce28ada1b11e10fbaea
http://b/issue?id=2972054
2010-09-03 10:19:23 +08:00
Chia-chi Yeh
dc296b0d4b
Merge "SipService: reduce the usage of javax.sdp.*." into gingerbread
2010-09-02 08:13:01 -07:00
Chia-chi Yeh
95b15c3560
SipService: reduce the usage of javax.sdp.*.
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After this change, SipAudioCallImpl is the only place still using it.
Change-Id: I5693bffa54f9e19cbfa70b45dfcf40fba04dedbb
2010-09-02 22:15:26 +08:00
Hung-ying Tyan
60264b3064
SipProfile: remove outgoingCallAllowed flag.
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Change-Id: I37a215bafce57adf6911c81fd38db324bac686ec
2010-09-02 20:34:17 +08:00
Hung-ying Tyan
3424c02e6b
Add software features for SIP and VOIP
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and block SipService creation and SIP API if the feature is not available.
Change-Id: Icf780af1ac20dda4d8180cea3e5b20e21a8350bc
2010-09-02 08:10:13 +08:00
Chung-yih Wang
0858806ffc
Add Wifi High Perf. mode during a call.
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To prevent the wifi from entering low-power mode due to the screen off
triggered by the proximity sensor.
Change-Id: I490bc594d800bc30c256e52ef3bce08bf86bc7b1
2010-08-26 15:05:48 +08:00
Chia-chi Yeh
14e00621c8
Merge "Revert "RTP: integrate the echo canceller from speex."" into gingerbread
2010-08-25 19:43:58 -07:00
Chia-chi Yeh
7fa7ee11f6
Revert "RTP: integrate the echo canceller from speex."
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This reverts commit 4ae6ec428f3570b9020b35ada6a62f94af66d888.
2010-08-26 10:33:09 +08:00
Chung-yih Wang
5424c8dcac
Add dynamic uid info for tracking the sip service usage.
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Change-Id: Ibc340401b63799326b08aee6eba602a3e753b13f
2010-08-26 10:12:05 +08:00
Hung-ying Tyan
37f709aeb0
Merge "SipProfile: add isOutgoingCallAllowed() and new builder constructor" into gingerbread
2010-08-24 23:57:50 -07:00
Hung-ying Tyan
cf95f5d263
SipProfile: add isOutgoingCallAllowed() and new builder constructor
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Change-Id: I7ced47079fd2b00c7160b152eb4c1d34399e39dc
2010-08-24 21:32:10 +08:00
Hung-ying Tyan
3294d44b96
Add confcall management to SIP calls
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and fix the bug of re-assigning connectTime's in SipConnection,
and adding synchronization for SipPhone to be thread-safe,
and set normal audio mode when call not on hold instead of on hold in SipAudioCallImpl,
and fix re-entrance problem in CallManager.setAudioMode() for in-call mode.
Change-Id: I54f39dab052062de1ce141e5358d892d30453a3a
2010-08-24 17:54:47 +08:00
Chia-chi Yeh
4ae6ec428f
RTP: integrate the echo canceller from speex.
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Currently the filter_length is set to one second.
Will change that when we have a better idea.
Change-Id: Ia942a8fff00b096de8ff0049a448816ea9a68068
2010-08-24 16:04:18 +08:00
Chia-chi Yeh
2880ef86e5
RTP: reduce the latency by overlapping AudioRecord and AudioTrack.
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Change-Id: I00d750ee514ef68d5b2a28bd1893417ed70ef1fc
2010-08-24 13:58:12 +08:00
Chia-chi Yeh
b879032347
RTP: fix few leaks when fail to add streams into a group.
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Change-Id: Iefb3fe219ad48641da37a83c8d14e9ebf1d3086c
2010-08-19 18:26:53 +08:00
Chia-chi Yeh
3459d3037c
RTP: remove froyo-compatible code.
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Change-Id: I6822a4e4749a5909959658c29253242b4018aeb0
2010-08-18 07:09:59 +08:00
Chung-yih Wang
cfd15dd3c8
Fix the IN_CALL mode issue.
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If the sip call is on-holding, we should not set the audio to
MODE_NORMAL, or it will affect the audio if there is an active pstn
call.
Change-Id: If1bcba952617bf8427bc9e2d64d483ba1ee37370
2010-08-16 18:02:31 +08:00
Hung-ying Tyan
ea4de5bd25
SipAudioCall: perform local ops before network op in endCall()
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Change-Id: I1808f715d56c0979cea7741cb5bdb3831774d3ef
2010-08-10 13:23:12 +08:00
Hung-ying Tyan
8e63ddb4c7
SIP: clean up unused class and fields.
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Change-Id: I79ed7fb324fea9a52946340055b5ea1d389a926a
2010-08-10 12:00:45 +08:00
Chia-chi Yeh
4c5d28cee0
RTP: move into frameworks.
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Change-Id: Ic9c17b460448c746b21526ac10b647f281ae48e9
2010-08-06 14:12:05 +08:00
Chung-yih Wang
cde66df442
Revert "Move SIP telephony related codes to framework."
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This reverts commit b631dcf3eb449ddec756bea330f4e70b996ffb9e.
2010-08-05 13:25:38 +08:00
Chung-yih Wang
b631dcf3eb
Move SIP telephony related codes to framework.
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+ hardcode the sip service for build dependency.
Change-Id: Ib0e9717c9b87eb6e06ffa3a7b01ae31184de61bb
2010-08-05 11:11:58 +08:00
Chung-yih Wang
363c2ab82c
Move the sip related codes to framework.
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Change-Id: Ib81dadc39b73325c8438f078c7251857a83834fe
2010-08-05 10:25:53 +08:00