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33 Commits

Author SHA1 Message Date
Chia-chi Yeh
37adc522f6 RTP: Add two getters to retrieve the current configuration from AudioStream. 
Change-Id: Iff588130653242f6ddd6a6b663df775ecb276768
 2010-09-23 03:34:14 +08:00
Chia-chi Yeh
32e106b7bd RTP: Extend codec capability and update the APIs. 
Change-Id: I37ba9d83c2de3c5dae2bfc1b7513df7f6fee3c5c
 2010-09-23 03:32:04 +08:00
Hung-ying Tyan
8544560ccc SipPhone: fix missing-call DisconnectCause feedback 
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. 
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. 
http://b/issue?id=2994029

Change-Id: I328da9b0f8b70d660dbcefffdac8250341792101
 2010-09-20 08:03:20 +08:00
Hung-ying Tyan
afa583e655 SipAudioCall: expose startAudio() 
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. 
+ 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. 
Change-Id: I839280fe18502bb576f6e9c9a7948077c02fa570
 2010-09-16 03:52:10 +08:00
Hung-ying Tyan
ae076d3981 SIP: add PEER_NOT_REACHABLE error feedback. 
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. 
+ 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() 
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 
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. 
+ 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. 
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. 
Change-Id: I77d289bef1b5e7f1ec0c0408d0bbf96c21085cd7
 2010-09-13 16:50:12 +08:00
Hung-ying Tyan
903e103160 SIP: add SipErrorCode for error feedback. 
Change-Id: I8b071d4933479b780a403d0bfa30511f4c23ca8f
 2010-09-10 17:15:06 +08:00
Hung-ying Tyan
643fce9781 SipManager: always return true for SIP API and VOIP support query. 
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.*. 
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. 
Change-Id: I37a215bafce57adf6911c81fd38db324bac686ec
 2010-09-02 20:34:17 +08:00
Hung-ying Tyan
3424c02e6b Add software features for SIP and VOIP 
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. 
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
Chung-yih Wang
5424c8dcac Add dynamic uid info for tracking the sip service usage. 
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 
Change-Id: I7ced47079fd2b00c7160b152eb4c1d34399e39dc
 2010-08-24 21:32:10 +08:00
Hung-ying Tyan
3294d44b96 Add confcall management to SIP calls 
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
b879032347 RTP: fix few leaks when fail to add streams into a group. 
Change-Id: Iefb3fe219ad48641da37a83c8d14e9ebf1d3086c
 2010-08-19 18:26:53 +08:00
Chung-yih Wang
cfd15dd3c8 Fix the IN_CALL mode issue. 
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() 
Change-Id: I1808f715d56c0979cea7741cb5bdb3831774d3ef
 2010-08-10 13:23:12 +08:00
Hung-ying Tyan
8e63ddb4c7 SIP: clean up unused class and fields. 
Change-Id: I79ed7fb324fea9a52946340055b5ea1d389a926a
 2010-08-10 12:00:45 +08:00
Chung-yih Wang
cde66df442 Revert "Move SIP telephony related codes to framework." 
This reverts commit b631dcf3eb449ddec756bea330f4e70b996ffb9e.
 2010-08-05 13:25:38 +08:00
Chung-yih Wang
b631dcf3eb Move SIP telephony related codes to framework. 
+ 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. 
Change-Id: Ib81dadc39b73325c8438f078c7251857a83834fe
 2010-08-05 10:25:53 +08:00