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272 Commits

Author SHA1 Message Date
Chia-chi Yeh
f4ae94229d RTP: Enable GSM-EFR codec. 
Change-Id: I9d84009e4557a0a82c1f9d7d543922741be97c77
 2010-09-30 03:07:57 +08:00
Chia-chi Yeh
fe5298992a RTP: Revise the workaround of private addresses and fix bugs. 
Change-Id: Ie654b569f47049aa452eca8d3e6d4a98ac18469c
 2010-09-30 02:43:48 +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
e006e4d2c9 Merge changes Iae1913fb,I38dbefef into gingerbread 
* changes:
  RTP: Enable GSM codec.
  RTP: Refactor out G711 codecs into another file.
 2010-09-28 19:40:59 -07:00
Chia-chi Yeh
a6f950c968 RTP: Enable GSM codec. 
Change-Id: Iae1913fb0643f1c66b5d16f24d51924d363e5ef5
 2010-09-29 10:36:52 +08:00
Chia-chi Yeh
9783052ec1 am df31e03c: am 320cdcb1: Merge "RTP: Delay the initialization of AudioTrack and AudioRecord." into gingerbread 
Merge commit 'df31e03c47cd4caf45d8a58cf1fe5893da48ec6c'

* commit 'df31e03c47cd4caf45d8a58cf1fe5893da48ec6c':
  RTP: Delay the initialization of AudioTrack and AudioRecord.
 2010-09-28 17:38:09 -07:00
Hung-ying Tyan
0b3968ae53 am 0d447760: am 6d028dd2: Merge "SIP: Feedback any provisional responses in addition to RING" into gingerbread 
Merge commit '0d44776016cecf1c7c826c4784f8f867a56235f0'

* commit '0d44776016cecf1c7c826c4784f8f867a56235f0':
  SIP: Feedback any provisional responses in addition to RING
 2010-09-28 17:37:04 -07:00
Chia-chi Yeh
78c11b3cf1 RTP: Refactor out G711 codecs into another file. 
Change-Id: I38dbefef2315a28d44683e86a51e69f38e3f20ec
 2010-09-29 05:46:19 +08:00
Chia-chi Yeh
320cdcb122 Merge "RTP: Delay the initialization of AudioTrack and AudioRecord." into gingerbread 2010-09-28 14:41:18 -07:00
Chia-chi Yeh
9083c84af1 RTP: Delay the initialization of AudioTrack and AudioRecord. 
Related to http://b/3043844.

Change-Id: I2c4fd9f64e6eba597d68b2ea1ceeff83103697db
 2010-09-29 05:24:34 +08:00
Hung-ying Tyan
6c6eacda80 am f7e13400: am 624d5b4e: SIP: add DisconnectCause.SERVER_ERROR 
Merge commit 'f7e13400b24bdc5461e640cfb2c9cb2bbb2b6236'

* commit 'f7e13400b24bdc5461e640cfb2c9cb2bbb2b6236':
  SIP: add DisconnectCause.SERVER_ERROR
 2010-09-28 12:47:49 -07:00
Hung-ying Tyan
6057cd00d9 SIP: Feedback any provisional responses in addition to RING 
The only exception is TRYING.
Also remove an unused import in SipSessionGroup.

http://b/issue?id=3021865

Change-Id: I160982b0c4b417362f1fb961217db90c3a585ce5
 2010-09-29 02:26:47 +08:00
Hung-ying Tyan
624d5b4e8c SIP: add DisconnectCause.SERVER_ERROR 
and fix how SipErrorCode.SERVER_ERROR is determinted from server response, not
from local exceptions.

http://b/issue?id=3041332

Change-Id: Idce67e29858d5c7573b98b7fa1fac074913d71d6
 2010-09-28 14:54:13 +08:00
Hung-ying Tyan
a57afb6a6c resolved conflicts for merge of 2a36a778 to master 
Change-Id: Ia70adeef06afddd29c827405fb5657bf9f5a29a3
 2010-09-28 12:17:44 +08:00
Hung-ying Tyan
7e54ef71db Move SipService out of SystemServer to phone process. 
Companion CL: https://android-git/g/#change,70187
http://b/issue?id=2998069

Change-Id: I90923ac522ef363a4e04292f652d413c5a1526ad
 2010-09-28 05:19:35 +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
Chung-yih Wang
5e18ad0c53 am 4a04a312: am bd229420: Fix the unhold issue especially if one is behind NAT. 
Merge commit '4a04a3129bd30a996dd302b982aeca8f228f57e8'

* commit '4a04a3129bd30a996dd302b982aeca8f228f57e8':
  Fix the unhold issue especially if one is behind NAT.
 2010-09-27 09:07:43 -07:00
Chung-yih Wang
bd2294204e Fix the unhold issue especially if one is behind NAT. 
+call startAudio() when call is established.

Change-Id: Ib6a1e34017fb83007ce275da1991058e8b803833
 2010-09-27 23:53:39 +08:00
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
repo sync
7a69aeffda RTP: Add log throttle for "no data". 
Change-Id: I14d9886a40fa780514cbc6c5bac6fb2a670f55f4
 2010-09-23 05:46:01 +08:00
Chia-chi Yeh
4033a67d0e RTP: Update native part to reflect the API change. 
Change-Id: Ic2858920ad77d7312f2429f89ca509a481363431
 2010-09-23 03:36:43 +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
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. 
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. 
Change-Id: I397a804e0aa598aee77a8ce28ada1b11e10fbaea
http://b/issue?id=2972054
 2010-09-03 10:19:23 +08:00