65a7f147deb02f728959eb05913a2d6ce53dea1c |
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11-Jan-2011 |
Hung-ying Tyan <tyanh@google.com> |
Get mute state from active call. Currently, PhoneUtils.getMute() returns the mute state from the foreground phone. When a SIP call is muted and then put on hold, the call is moved to background and the SipPhone becomes background phone. At this point, PhoneUtils.getMute() incorrectly returns false from the idle foreground phone (i.e., GSMPhone). CallManager provides getMute() but it's not used anywhere. This CL fixes the method and I'll have another CL to have PhoneUtils.getMute() take advantage of it. Bug: 3323789 Change-Id: I6c37500ae93f4e95db3bcd55e24e1ecb58a57c0a
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
1d12ef09a8e6ebc6638f4ff2f561c50c950023cb |
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13-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
Fix setting audio group mode in SipPhone. Bug: 3119690 Change-Id: I495d3c031ee4c272d360fe19553ef9726a3f8771
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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f053292d7a46c30abbe6f12ca04dbc03ec964d80 |
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03-Nov-2010 |
Chung-yih Wang <cywang@google.com> |
Fix SIP bug of different transport/port used for requests. bug: http://b/3156148 Change-Id: I4fa5b274d2e90ebde12d9e99822dc193a65bad32
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
4189d99b6e4877352049b7447b7f0734ef99b9e8 |
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24-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Do not suppress error feedback during a SIP call. Bug: 3124788 Change-Id: Ia0a06f72336d1795515428eba0c9f875c32d13d1
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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8d1b2a17d9935819ec96f1b5fca0e9945f564eaa |
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03-Nov-2010 |
Hung-ying Tyan <tyanh@google.com> |
Throw proper exceptions in SipManager instead of silently returning null and causing NPE in applications as returning null is not documented in the javadoc. Add connection to the connection list in SipCall after dial() succeeds so that we don't need to clean up if it fails. The original code will cause the failed connection to continue to live in the SipCall and in next dial() attempt, a new connection is created and the in-call screen sees two connections in the call and thus shows conference call UI. Bug: 3157234, 3157387 Change-Id: Iabc3235f781c4f1e09384a67ad56b09ad2c12e5e
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
6037a056ea0dda27a286ddcb527b323b58a1c7c7 |
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20-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Fix n-way conf call in SipPhone. + Avoid concurrent modification when forming >3-way conf call. + Revise SipConnection.separate() to put the newly separated call to foreground. Bug: 3114987 Change-Id: If6204e7e3cc05f4a516c33657a368b53a0ad014d
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
9b449e5606786f7c197679f8f9d25985308bfb72 |
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20-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Remove ringtone API from SipAudioCall. (watch out auto-merge conflict for SipAudioCall). Bug: 3113033, related CL: https://android-git/g/#change,75185 Change-Id: Ib48d3b990e229e0b341e47e10e76934e1a50d10f
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
538e58fc757b0d10672235bc17b1380854845139 |
|
20-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Return display name in SipConnection.getCnapName(). Bug: 3105116 (case #1) Change-Id: Iedf3c8de07213c786cffb861bd52c3b4a768a86c
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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f5201ab71ff4d104265ab126e86afc6b81da8011 |
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12-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Keep original phone number in SipConnection. In case it's a PSTN number carried by an Internet call, the phone app can still get the original phone number from Connection.getAddress() instead of getting a SIP URI. http://b/issue?id=3085996 Change-Id: Ie6c66100a4b5b2ce3f73baa1b446761cd51d7727
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
d07833f54b6e8e361b666ae16efa15fdf60159de |
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08-Oct-2010 |
David Brown <dab@google.com> |
Don't manually create CallerInfo objects from SipPhone Currently the SipPhone class manually creates a CallerInfo object, and populates it with very basic info from the SIP address, when making an outgoing call. But this is no longer needed, now that we do caller-id lookup properly for SIP addresses (based on real data from the contacts database -- see bug 3004127 and change https://android-git.corp.google.com/g/70555). And in fact the presence of this initial CallerInfo object actually *disabled* contacts lookup for outgoing calls (bug 3072731). This change removes all that CallerInfo-related stuff from SipPhone. (Thus SipPhone is now consistent with the other phone objects, like GSMPhone and CDMAPhone, in that it doesn't muck with CallerInfo data at all, but instead lets the phone app do it.) Also, update isUriNumber() to handle "%40" in case the passed-in string is URI-escaped. (Nobody depends on that now, but it may be needed in the future, and it's certainly safe to say that "%40" will never be found in a legal PSTN number.) TESTED: - Outgoing SIP call: - In-call UI shows correct contact info - After the call, Call Log shows correct contact info - Incoming SIP call: - In-call UI shows correct contact info - After the call, Call Log shows correct contact info - PSTN calls: - correct contact info everywhere Bug: 3072731 Change-Id: I51434e4e5ad66d2e8ff51fc220001fb74485f0f5
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
17956e626b38ce53da61e78af2c973ed41c9e461 |
|
01-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
CallManager/SipPhone: fix reject a ringing call + CallManager: fix getFirstActiveRingingCall(), getActiveFgCall(), getFirstActiveBgCall() + Set DisconnectCause to be INCOMING_REJECTED when a call is rejected http://b/issue?id=3049671 Change-Id: Ica1d81ca4b71ab0ceb2ab437b82bbb4ccf86fe92
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
c6548fd9eda7b58f5a2e2a9c01e3c7cafd42fafb |
|
05-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: add SERVER_UNREACHABLE error code. Let SipSession return it when UnknownHostException is caught. Add DisconnectCause.SERVER_UNREACHABLE in Connection and have SipPhone report it when receiving SERVER_UNREACHABLE from SipSession. http://b/issue?id=3061691 Change-Id: I944328ba3ee30c0a9386e89b5c4696d4d9bde000
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
306137d97f40a4f807c54a75210343c9262360d1 |
|
01-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP telephony cleanup. + Remove unused classes. + Remove unused imports. + Remove unused code. + add DEBUG flag. Change-Id: Ie1236d909d971093b68b066d3d8c1857ac89f56f
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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23f21600d0927365e5e7bdc4e566ba52101301b4 |
|
29-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Add CallManager/Phone.setEchoSuppressionEnabled(). Change-Id: I7bc6241e6fa815787799a53d6f3a076567edc361
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
0e412304813ccd3a3fb6a643836e4f0922d1dc44 |
|
28-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Catch exceptions in SipPhone.canTake(). Exceptions may throw during canTake() as the peer may cancel the call and result in a race with this method call. Change-Id: I61903d601d8f9b2dcb4c4fbe1586e2c1a1069109 http://b/issue?id=3033868
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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421c34c162098efe870574844a7ee49812bbb929 |
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28-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipPhone: revise hangup() in SipCall and SipConnection. Make them DISCONNECTED immediately. Don't enter DISCONNECTING state and wait until SipSession ends the session. SipSession will get timed out eventually but PhoneApp/user don't need to know this detail and wait. This should fix the bug: http://b/issue?id=3027719 Change-Id: Ida5a1bd09d08b9d591721384b4978127619aab51
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
624d5b4e8c20516516d0bff74479b9f5abdfe61c |
|
28-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
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
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
245475925eff61ee76bde58de69253a889e39d0a |
|
28-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
Fix the startAudio order for 3-way calls. Change-Id: Ib387b4b1f641f9bf52dd6007d23aee08f0925811
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
368d796e2e28ecd738362c7a4566cb3eb219ab26 |
|
27-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Fix build. Change-Id: I30f2615bc080db2c672e0391fd8bc735de17fcbf
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
2b4f5cfd9be5ceffc4745a45736e067a475a4dff |
|
28-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Don't enter DISCONNECTING state when the call/connection is not alive http://b/issue?id=3027719 Change-Id: I1b52418a3695e96b48538fbf14497e34d2cfdda9
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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025a39af346f39743c1e384b9000ce1baee36562 |
|
23-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: misc fixes + check REQUEST_TERMINATED response on INVITE not CANCEL, + check if a TransactionTerminatedEvent matches the ongoing transaction, + add log to track SipConnection disconnect events. Change-Id: I28325be62ac44e4a7507d3c4b5b78b066c0ea2ad
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
00a22064efef4f574e439079aae2deae1a087a31 |
|
25-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
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
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
bd2294204e3edaede3fe81eb9b11c05c4fafe627 |
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23-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
Fix the unhold issue especially if one is behind NAT. +call startAudio() when call is established. Change-Id: Ib6a1e34017fb83007ce275da1991058e8b803833
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
194bbcce9ba15634500f542b9ea017b2cf154b45 |
|
23-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: longer timeout for making call, shorter for cancelling http://b/3021865 Change-Id: I354ebcc00f1ac68e4b7b466745c36aeb314f9138
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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84a357bb6a8005e1c5e924e96a8ecf310e77c47c |
|
15-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
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
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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9779b714f4035642b87cbb7ef6cd8ac32848c930 |
|
19-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
Revert the ANSWERING state. +fix the unknown call flash for answering an incoming call and updating the screen if the background call got dropped. +change the getFirstActiveBgCall to return the call if the state is not IDLE. This will help to fix unknown flash if the background call got dropped. Change-Id: I9803ccebd919acbd5296e7dfde7dc5f29cc9f180
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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8544560ccc43de7ff49d91866f461f5572f0b147 |
|
20-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
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
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
97963794af1e18674dd111e3ad344d90b16c922c |
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17-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: convert enum to static final int. Converts SipErrorCode and SipSessionState. Change-Id: Iee3a465649ea89d395b2336bbd673c25113e5f93
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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1d1583573d2099756bbbeef48d97c280edc393e0 |
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17-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipPhone: do not append SIP domain to PSTN number in the CallerInfo so that only PSTN number is shown in the call log. http://b/issue?id=2982632 Change-Id: I414f01d16ce64ecb8da7c6943ea7f080bcfd2794
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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afa583e6557557577188c3e40146ac8d6f2aa7c7 |
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17-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipAudioCall: expose startAudio() so that apps can start audio when time is right. Change-Id: I7ae96689d3a8006b34097533bc2434bc3814b82a
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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9352cf1a4d46492fc48a20f7d825a9bcb6e8b365 |
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16-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
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
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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d8f3d167353f6c6f6c5cb7a4c8e941c03b8e9511 |
|
16-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
Add a new phone state ANSWERING. The state ANSWERING is set when we answer an incoming sip call, i.e. sending a 'OK' response to the peer. The state will be set to ACTIVE once the 'ACK' from peer is received. Change-Id: I84ee3cc68222eb34e032896ce23f7431d4ad774a
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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94e498332a4e114dd106f564ebdafb49acea9854 |
|
15-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
Fixing the NPE in SipPhone bug id: http://b/2987816 Change-Id: Iee252eee0a5243b70ff0b6f287279f92235b5b2d
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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ae076d3981fda732d54b6c6e37e5659b2e7ba130 |
|
15-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: add PEER_NOT_REACHABLE error feedback. http://b/issue?id=3002033 Change-Id: Ib64b08919d214acbab89945ac19dc113a68e62ad
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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13f6270eb14b409709c936b828e2a2fd40e427c4 |
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14-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipAudioCall: use SipErrorCode instead of string in onError() and fix callback in setListener(). Change-Id: Ic2622df992a2ad45cb1e3f71736f320897ae8fb3
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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d231aa880ab006d51ffe03454c1fc082f1c97bb8 |
|
13-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
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
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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3d7606aa607b24817e37c264f2141ed7b2d50be0 |
|
12-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: enhance timeout and registration status feedback. http://b/issue?id=2984419 http://b/issue?id=2991065 Change-Id: I2d3b1dd3a70079ff347f7256f4684aea07847f4e
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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25b52a2f97df112c2836972d0b6d9a4c7a9c4a4e |
|
13-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: remove dependency on javax.sip.SipException. Change-Id: I77d289bef1b5e7f1ec0c0408d0bbf96c21085cd7
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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903e1031605d715e904811b0dd06cc6a518f0048 |
|
09-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: add SipErrorCode for error feedback. Change-Id: I8b071d4933479b780a403d0bfa30511f4c23ca8f
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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b6264a8795ed9469c80727123e3cafda1b07eda3 |
|
05-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
Fix the build. Change-Id: Icfeec3372dcde30723c49565649be03a4dd33c06
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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b12baad9357c6e6aec1f7d84fd041c54fe963407 |
|
06-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
Add equals() funcation for SipPhone. Since we will use sipuri to match the same phone object. Change-Id: I582779e51e447bb8d822c105cf0d682651c138d2
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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3294d44b96f63f647fba3a03604eb028e28a42bc |
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18-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
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
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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8eac20eacd088793547c56e14d602b28d62fb278 |
|
17-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: implement conference call Change-Id: Ifd420ed95e77e744c6aff28ac63e7363f97d9dc6
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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f57324cf4f82947296f4d1acb9df1f3c9c03134e |
|
11-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Add getSipUri() for identification. Change-Id: Iabffd38ad554c34a34977c833e6699747cbf0f63
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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8e63ddb4c78dc4453d64ea6e94c109db703185e4 |
|
09-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: clean up unused class and fields. Change-Id: I79ed7fb324fea9a52946340055b5ea1d389a926a
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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ccd0b6953f5f77d1da5f540a3ba5ea71116e14f0 |
|
05-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Revert "Revert "Move SIP telephony related codes to framework."" This reverts commit cde66df44240cfe5a7bec12ac52464c3bf26c14f. Change-Id: I87da883b45350ec8f7da71e9bd392b075ea30ca7
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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cde66df44240cfe5a7bec12ac52464c3bf26c14f |
|
05-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Revert "Move SIP telephony related codes to framework." This reverts commit b631dcf3eb449ddec756bea330f4e70b996ffb9e.
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
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b631dcf3eb449ddec756bea330f4e70b996ffb9e |
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05-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Move SIP telephony related codes to framework. + hardcode the sip service for build dependency. Change-Id: Ib0e9717c9b87eb6e06ffa3a7b01ae31184de61bb
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|