06a98183cc79dd112d9d33cf027977a9d5d3418a |
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02-Nov-2016 |
Tyler Gunn <tgunn@google.com> |
Catch SIP exceptions which can crash Phone process on answer. There are two exceptions which can be raised when answering a call which can cause the Phone process to crash on answer. 1. IllegalStateException due to answering a call with an incompatible codec. 2. IllegalArgumentException due to answering a call with a malformed SDP. In both of these cases we catch the exception and reject the call to stop it from ringing (otherwise it will keep ringing and the user will not be able to stop it). The existing CallStateException does not require onReject to be called as it is thrown when the call has already been disconnected before it can be answered. Test: Manual (see bug) Bug: 31752213 Change-Id: I5254fd3a27b86fdc70889ea0a2b5be3b699fd9f5
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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7877c7255268c34a71c7d1b4055afabb8fcba04c |
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07-Mar-2016 |
Hall Liu <hallliu@google.com> |
Add the call technology extra for SIP calls for analytics Set the EXTRA_CALL_TECHNOLOGY_TYPE extra on SipConnection. Bug: 27529579 Change-Id: I2288057f25a31ef04a716b9f11ac18ed38509ae3
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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3ba1419ade788f717dc2beb9667224dc9ed2e317 |
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16-Sep-2015 |
Santos Cordon <santoscordon@google.com> |
Double-check SIP state before issuing a swap-calls command. Our code relied on the state of SipConnection to prevent us from holding a HELD call and unholding an ACTIVE call. However, while the action is in progress, the SipConnection state can be slightly delayed so we double-check the internal state of the lower connection prior to issuing a switchHoldingAndActive command. Bug: 24007856 Change-Id: I617ef215e66f6b9324dc17cb1b9832cc35390f17
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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caf84dca8bd90016d910198f1b178ac03e82dd07 |
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18-Sep-2015 |
Roshan Pius <rpius@google.com> |
Disallow call unhold if the fg call is dialing. When a SIP call is in dialing state, it cannot be held. Hence ignore switching of fg/bg calls when there are multiple calls and the foreground call is still in dialing state. This would prevent the user from unholding the background call and hitting a crash when we try to switch out the foreground dialing call. BUG: 17448699 Change-Id: I816d3c3f0253811f1ae4e7aeaa138a830d0c4d25
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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2c304ad93f8e74e4a14374d6e00a50cd3497d592 |
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21-May-2015 |
Kosuge Yuji <yuji.x.kosuge@sonymobile.com> |
Two calls screen is not displayed on multiple SIP calls When accepting the 2nd incoming SIP call, the 1st SIP call is set to Hold call by calling SipPhone#switchHoldingAndActive(). However, Hold call is returned to Active call afterwards because switchHoldingAndActive() is called by the TelephonyFWK side again. Therefore, two calls screen is not displayed because Hold call does not exist. Change-Id: I623a67c2493fe7c4bcd8b36f3b13ede31f36b820
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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3cc8121590b5b05af65a9aeaa8948834e4983adc |
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12-Nov-2014 |
Ihab Awad <ihab@google.com> |
Telecom API updates (5/6) Bug: 18292176 Change-Id: I706c5a120e8a136d084d4b7f9f0fbc2653a5b86b
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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aef7a4bc4f85149de427d7506ebe97753b2ca6c2 |
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12-Sep-2014 |
Andrew Lee <anwlee@google.com> |
Use newly added Telecomm DisconnectCase. + Add utility, DisconnectCauseUtil, to convert from telephony DisconnectCauses to telecomm DisconnectCauses. For this, map from specific disconnect causes into a more generic set which the UI is concerned about. Provide user-localized strings or tones for certain cases. - The string conversion sucks a little because it covers both strings which are intended to be show in the InCallUi (short) and messages which are to be displayed to the user in a dialog when there is some termination (long). =( + Where possibly, use this utility for converting error messages as well. - Not able to convert with message and tone for many of the failure cases, because no context exists for obtaining the message to display. Bug: 17329632 Change-Id: I84ee358cc8e014f110bab4c0ae7c1cb02aeebdf6
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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4d45d1cf58a2003378fd35912d6d73a00001bf06 |
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13-Sep-2014 |
Tyler Gunn <tgunn@google.com> |
Renaming Telecomm to Telecom. - Changing package from android.telecomm to android.telecom - Changing package from com.android.telecomm to com.android.server.telecomm. - Renaming TelecommManager to TelecomManager. Bug: 17364651 Change-Id: Ic33bafd37200e65431543a9889aed549dbb04831
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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7ccf4d2bff43392dd7db21e12e17400ae7aa87fc |
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11-Sep-2014 |
Andrew Lee <anwlee@google.com> |
Merge "Rename from "handle to address"." into lmp-dev
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eeff153f4cbeea5d4d1ce27c2f9ca2ea0ca482c8 |
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11-Sep-2014 |
Nancy Chen <nancychen@google.com> |
Merge "Make changes to Connection in Telecomm API (2/4)" into lmp-dev
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14bf354aab420fcd099cd0209238d7b92dad6f1d |
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09-Sep-2014 |
Andrew Lee <anwlee@google.com> |
Rename from "handle to address". Bug: 17329632 Change-Id: Ib947b16cffee343bc7ca36d03617d1f7e4b3bd1c
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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a314b4f398ec4a0f1540c9e437bf2d7f99502b6a |
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10-Sep-2014 |
Anthony Lee <anthonylee@google.com> |
Make sure that we request a ringback on SIP connections. Bug: 17383445 Change-Id: Ie8399453a381216e22087362e37703acbd96f311
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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c927106af6c47ad1f302e13fa8626be83bb46f14 |
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09-Sep-2014 |
Nancy Chen <nancychen@google.com> |
Make changes to Connection in Telecomm API (2/4) * onPhoneAccountClicked removed * onSetAudioState -> onAudioStateChanged * onSetState -> onStateChanged Bug: 17329632 Change-Id: If070818eaed0372c37cfbc20733cf8206a67cca8
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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137458b4bf3516941483e59c123c22cbee27ed43 |
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05-Sep-2014 |
Jay Shrauner <shrauner@google.com> |
Use framework scheme definitions Use PhoneAccount defined values for SCHEME_{TEL, SIP, VOICEMAIL}. Bug:17398074 Change-Id: I36adb16f659daef89957072f9e00d08ea2cb8e9e
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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4ddd0efe67e93798d98c86b073d6cd0b1343f840 |
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02-Sep-2014 |
Tyler Gunn <tgunn@google.com> |
Add supported URI scheme to PhoneAccounts. (3/4) 1. Modify TelephonyConnectionService to register voidemail URI scheme. 2. Modify SipConnectionService to register 'tel' URI scheme in addition to 'sip', should the user have chosen to use sip for all calls. 3. Modify SipBroadcastReceiver to listen to ACTION_SIP_CALL_OPTION_CHANGED intent, triggering a rebuild of the SIP PhoneAccounts. 4. Fixed bug where all incoming SIP calls show as Unknown number. Bug: 17140110 Bug: 17326799 Change-Id: I11d25c2cd98d02b80919d95ae9af077edb179fe0
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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6556a09daab949853c384b385bc7618a6c75d9dd |
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25-Aug-2014 |
Santos Cordon <santoscordon@google.com> |
Add telecomm registration support to SIP. Changes in CL: 1. Add SipAccountRegistry to manage telecom registrations 2. Register all accounts on ACTION_SIP_SERVICE_UP 3. Register specific account on ACTION_SIP_ADD_PHONE 4. Unregister specific account on ACTION_SIP_REMOVE_PHONE 5. Set the unique SIP uri as the PhoneAccount ID 6. Read phone account handle for outgoing calls instead of showing a proprietary SIP chooser dialog, which I removed (see SipProfileChooser.java). 7. Moved some of the error condition codes and error dialogs from SipProfileChooser to SipConnectionService. 8. Set Phone account handle as extra on the incoming-call intent 9. Remove build files for SIP directory since it is already built into TeleService.apk Bug: 16836473 Bug: 16042786 Change-Id: I62d740c13ce61f181db295b9415c96ceef909177
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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53b84fe2dc796ef172d7c0f4b9bdc177cdeb0c0f |
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13-Aug-2014 |
Santos Cordon <santoscordon@google.com> |
Implement new Conference APIs. Bug:16844332 Bug:16449372 Change-Id: Id2e1e4996c19ca1fa4f37e1ec6597f3a15676aa8
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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76f3b4ec8d9bcb8926db1b3e4fb2d1e969b09fbb |
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08-Aug-2014 |
Ihab Awad <ihab@google.com> |
Final structural tweaks to Telecomm API (5/8) Bug: 16416927 Bug: 16494880 Change-Id: Ibc2ba9e5d17242380ed3afed359600362bb3e664
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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da120f4e3d32ca97c5b4c21d6c505d834a29ab8d |
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06-Aug-2014 |
Santos Cordon <santoscordon@google.com> |
Remove ldaps from AOSP code. Bug: 16327484 Change-Id: Ic51f12e52db0839a9ee6b727755873edf65cd044
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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aaf3850172bde1fec1a4558653c0edd31e3fa71f |
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21-Jul-2014 |
Evan Charlton <evanc@google.com> |
Use the synchronous Connection creation API Update to make creating a Connection a synchronous operation. Change-Id: Ibbf896e5b188b98eb386d6624d3a5d5cacf49922
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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ab777070da7e83983739414f1222177c6aeebe1a |
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21-Jul-2014 |
Santos Cordon <santoscordon@google.com> |
Change telecomm APIs from protected to public (3/3) Bug: 16416927 Change-Id: I859c80a120fa59bba12a5c6bfca17fa57613a6aa
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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0d5710d9bfdb36d5d0a7be4eaf3f39ae3fb30bc5 |
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20-Jul-2014 |
Evan Charlton <evanc@google.com> |
Rename setDestroy() to destroy() Change-Id: I6bb46e679f55547b902d5850ab450e5d3818fa63
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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0bbcc4f2f0218f451a680b98a62c374bba877c3b |
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16-Jul-2014 |
Andrew Lee <anwlee@google.com> |
Adding videoState parameter to acceptCall method on Connection. Bug: 16013878 Bug: 16015750 Change-Id: Ib7d4274ff958157581de000713cb114bac789cc0
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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22d32b2b81bba31b6ebd93753a5c1ec39e911981 |
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15-Jul-2014 |
Sailesh Nepal <sail@google.com> |
Remote incoming calls: Telephony Change-Id: Ie8091aacfd9be36d63db02eb8a0a4935753d657f
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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2093a451b17c26f4341e9565b65dcaa0e20bbd7d |
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12-Jul-2014 |
Sailesh Nepal <sail@google.com> |
Mis Telecomm API changes - telephony This CL fixes up Telephony to work with the API changes in ag/501321. Change-Id: Iaffe0b84cea6003f2a9b9d8b30676743d2b236d4
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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1344f67ea331f9a485f54c4b5e26d62a5cfad3fb |
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11-Jul-2014 |
Sailesh Nepal <sail@google.com> |
Hookup incoming and outgoing SIP calls This CL hooks up SipConnection and SipConnectionService. Bug: 14999064 Change-Id: Iddf4a06c70fb73398844e48a80be4c63f6b82e73
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
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788959e2d798da2d8a34cf89779421966d200f3d |
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09-Jul-2014 |
Sailesh Nepal <sail@google.com> |
SIP Part 1, move sip code This CL moves the SIP code to sip/src/com/android/services/telephony/sip Moving the SIP code helps with a couple of things: - remove legacy dependency on old telephony code (PhoneUtils, etc...) - separate SIP from the PSTN connection code which is very different from SIP code - get the code ready for a future move out of Telephony all together Bug: 14999064 Change-Id: Id32de6517d31be4aa177b2764d5bac1e1f9851c2
/packages/services/Telephony/sip/src/com/android/services/telephony/sip/SipConnection.java
|