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106 Commits

Author SHA1 Message Date
Chia-chi Yeh
a77c9541d0 am cbee6229: am 0a537b78: Merge "RTP: Enable AMR codec." into gingerbread 
Merge commit 'cbee622954de5e9e0c07557f8ec9aaa741110043'

* commit 'cbee622954de5e9e0c07557f8ec9aaa741110043':
  RTP: Enable AMR codec.
 2010-09-30 12:02:10 -07:00
Hung-ying Tyan
d161479237 am 947d2abd: am 2365b78e: Merge "SIP: misc fixes." into gingerbread 
Merge commit '947d2abd82ef68c661fc29fd5167e4c0ba749f71'

* commit '947d2abd82ef68c661fc29fd5167e4c0ba749f71':
  SIP: misc fixes.
 2010-09-30 12:02:03 -07:00
Chia-chi Yeh
5a7c6d298e am 1c2eab2d: am 955ab37c: Merge "RTP: Enable GSM-EFR codec." into gingerbread 
Merge commit '1c2eab2d86faa9c647a9893f761a50cfa28d9d57'

* commit '1c2eab2d86faa9c647a9893f761a50cfa28d9d57':
  RTP: Enable GSM-EFR codec.
 2010-09-30 10:59:48 -07:00
Hung-ying Tyan
9e1d308e99 Add uri field to SipManager.ListenerRelay 
in case mSession is not available.

Change-Id: Ifee2c129e48aa1177f648f176413ab6aa5606770
 2010-09-30 15:00:34 +08:00
Chia-chi Yeh
0a537b78d3 Merge "RTP: Enable AMR codec." into gingerbread 2010-09-29 18:32:24 -07:00
Hung-ying Tyan
2365b78e64 Merge "SIP: misc fixes." into gingerbread 2010-09-29 18:12:12 -07:00
Chia-chi Yeh
f88fc1fa90 RTP: Enable AMR codec. 
Change-Id: I49e6bdc1b67306b44173f2f346f8372a50264870
 2010-09-30 08:55:12 +08:00
Hung-ying Tyan
fb3a98b1d8 SIP: misc fixes. 
+ Fix keepalive timer event leak due to the race between stopping timer and
  the async'ed timeout handler
+ SipSessionImpl: set state before handling an event to ensure we get correct
  state when some error occurs during handling the event.
+ Fix potential NPE in SipManager.ListenerRelay.getUri().

Change-Id: I021ee34f83059fd4fbb64b30bea427a5462aa51b
 2010-09-30 08:10:17 +08:00
Chia-chi Yeh
f4ae94229d RTP: Enable GSM-EFR codec. 
Change-Id: I9d84009e4557a0a82c1f9d7d543922741be97c77
 2010-09-30 03:07:57 +08:00
Chia-chi Yeh
dcf2be6cf6 am ebfe5632: am e006e4d2: Merge changes Iae1913fb,I38dbefef into gingerbread 
Merge commit 'ebfe5632db275a89b49ab828064ba90db59702cf'

* commit 'ebfe5632db275a89b49ab828064ba90db59702cf':
  RTP: Enable GSM codec.
  RTP: Refactor out G711 codecs into another file.
 2010-09-28 19:47:07 -07:00
Chia-chi Yeh
a6f950c968 RTP: Enable GSM codec. 
Change-Id: Iae1913fb0643f1c66b5d16f24d51924d363e5ef5
 2010-09-29 10:36:52 +08:00
Hung-ying Tyan
5a474a2bb8 am 44669d31: am fd144d76: Merge "SipAudioCall: remove SipManager dependency." into gingerbread 
Merge commit '44669d31d1d5b094d7b7d3e393281440ea0c9536'

* commit '44669d31d1d5b094d7b7d3e393281440ea0c9536':
  SipAudioCall: remove SipManager dependency.
 2010-09-27 11:47:42 -07:00
Hung-ying Tyan
031d878682 am fe2d279c: am 00a22064: SipService: handle cross-domain authentication error 
Merge commit 'fe2d279c5ef571340f20d433badd9f68072299af'

* commit 'fe2d279c5ef571340f20d433badd9f68072299af':
  SipService: handle cross-domain authentication error
 2010-09-27 11:47:32 -07:00
Hung-ying Tyan
fd144d7667 Merge "SipAudioCall: remove SipManager dependency." into gingerbread 2010-09-27 10:54:27 -07:00
Hung-ying Tyan
00a22064ef SipService: handle cross-domain authentication error 
and add new CROSS_DOMAIN_AUTHENTICATION error code and OUT_OF_NETWORK
DisconnectCause.

http://b/issue?id=3020185

Change-Id: Icc0a341599d5a72b7cb2d43675fbddc516544978
 2010-09-27 10:45:24 -07:00
Hung-ying Tyan
3a4197e642 SipAudioCall: remove SipManager dependency. 
Change-Id: I2dc8bf427e52f64529ee0e0261362b975a8917c6
 2010-09-24 23:27:40 +08:00
Hung-ying Tyan
a97c5f7779 Merge "fix build" 2010-09-23 23:24:35 -07:00
Hung-ying Tyan
fb0264096e fix build 
Change-Id: Iff05b5ea7f535f532eec2af1edf78fdf8acfa21c
 2010-09-24 14:23:31 +08:00
Chia-chi Yeh
658bec9567 SDP: remove dead code. 
Change-Id: I2a5764a2b9cabc54b0ac18666e494c1cb39c4e9b
 2010-09-24 10:17:42 +08:00
Hung-ying Tyan
84a357bb6a Refactoring SIP classes to get ready for API review. 
+ replace SipAudioCall and its Listener interfaces with real implementations,
  + remove SipAudioCallImpl.java, most of it is has become part of SipAudioCall,
+ add SipSession and its Listener classes to wrap ISipSession and ISipSessionListener,
+ move SipSessionState to SipSession.State,
+ make SipManager keep context and remove the context argument from many methods of its,
+ rename SipManager.getInstance() to newInstance(),
+ rename constant names for action strings and extra keys to follow conventions,
+ set thread names for debugging purpose.

Change-Id: Ie1790dc0e8f49c06c7fc80d33fec0f673a9c3044
 2010-09-24 10:06:59 +08:00
repo sync
0b7d6de155 Fix the build. 
Change-Id: I82210cb2d41f532583f83ea17e6f2d8d49280a30
 2010-09-23 14:52:24 +08:00
repo sync
84f7f6ba39 SIP: Make SipAudioCallImpl use SimpleSessionDescription instead of javax.sdp. 
Change-Id: I7efff4f29ca84c3e7c17ef066b7186b514a777b2
 2010-09-23 14:07:45 +08:00
Chia-chi Yeh
e6c0c10958 SDP: Add a simple class to help manipulate session descriptions. 
Change-Id: I1631ee20e8b4a9ad8e2184356b5d13de66e03db1
 2010-09-23 13:31:01 +08:00
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