| fae5e2894ff3c09f27efac2a7ee6b9cfd4ed14b0 |
|
29-Mar-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am 6f67e7bf: am 2e383bc6: Merge "Making it possible to call SIP calls with special allowed chars."
* commit '6f67e7bf831147257e078dd72a22f2e43e009122':
Making it possible to call SIP calls with special allowed chars.
|
| b5c72ead014a509c0f84884d1f2dac1ff9deec8e |
|
22-Mar-2011 |
Magnus Strandberg <magnus.strandberg@sonyericsson.com> |
Making it possible to call SIP calls with special allowed chars.
Since String.replaceFirst uses regex and since SIP user names are
allowed to include regex charaters such as '+', the code must
fist convert the string to a literal pattern String before using
replaceFirst method.
Change-Id: I25eac852bd620724ca1c5b2befc023af9dae3c1a
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
| b30392d288c6c72a6db00a71a2ca586793161b48 |
|
25-Jan-2011 |
Hung-ying Tyan <tyanh@google.com> |
am df1cc4ef: am cc019c0c: Merge "Get mute state from active call." into gingerbread
* commit 'df1cc4ef9205239701bbe73f903e89a9dfd2623a':
Get mute state from active call.
|
| 65a7f147deb02f728959eb05913a2d6ce53dea1c |
|
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
|
| 8fd2718bf3dfd67a26619a1ce765f8423c8c4bfe |
|
05-Jan-2011 |
Hung-ying Tyan <tyanh@google.com> |
am 05c53067: am 273d2ea3: Merge "Fix setting audio group mode in SipPhone." into gingerbread
* commit '05c53067b613796624937214e506c58de817c8fd':
Fix setting audio group mode in SipPhone.
|
| 1d12ef09a8e6ebc6638f4ff2f561c50c950023cb |
|
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
|
| c030a164c8a890947985d15722fe3df8785f7d04 |
|
07-Dec-2010 |
Chung-yih Wang <cywang@google.com> |
am c9cc9ab5: am 5f86d7f5: Merge "Fix SIP bug of different transport/port used for requests." into gingerbread
* commit 'c9cc9ab590ef879877e466c0b5f5823e11bb4c47':
Fix SIP bug of different transport/port used for requests.
|
| f053292d7a46c30abbe6f12ca04dbc03ec964d80 |
|
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
|
| db4245291b15fd966b36c70f7f69ba4d22539803 |
|
01-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
am ed34b244: am d7116ff1: Merge "Do not suppress error feedback during a SIP call." into gingerbread
* commit 'ed34b244f1665b604d2a291db504415b10a514d7':
Do not suppress error feedback during a SIP call.
|
| ebf28fa3f086bd5d3fa8d988fe4b8a8faeddd710 |
|
01-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 0e58a952: am 0bba9535: Merge "Throw proper exceptions in SipManager" into gingerbread
* commit '0e58a9529895e270dae90e69486a59e41de714b8':
Throw proper exceptions in SipManager
|
| 4189d99b6e4877352049b7447b7f0734ef99b9e8 |
|
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
|
| 8d1b2a17d9935819ec96f1b5fca0e9945f564eaa |
|
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
|
| 2cef210f53ac9ca5471e87fa02db252442448b7d |
|
21-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 940b4d1c: am 6037a056: Fix n-way conf call in SipPhone.
Merge commit '940b4d1c4548d3296ac9fc66cce0cc213b5aa8a8'
* commit '940b4d1c4548d3296ac9fc66cce0cc213b5aa8a8':
Fix n-way conf call in SipPhone.
|
| 164cd438fb21e82d0aacc06da940041f0b7f6a2c |
|
21-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 51028569: am 1180f2a0: Merge "Remove ringtone API from SipAudioCall." into gingerbread
Merge commit '5102856947595cffc1cceb11b9e4c5baf70b2e82'
* commit '5102856947595cffc1cceb11b9e4c5baf70b2e82':
Remove ringtone API from SipAudioCall.
|
| 4e00a8d5826c5a81d5c9b4a5f6c4ee40e00e5426 |
|
21-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am e894ff7a: am b595e094: Merge "Return display name in SipConnection.getCnapName()." into gingerbread
Merge commit 'e894ff7a7693fdc406b4f4b28cfd1d9d7d966b38'
* commit 'e894ff7a7693fdc406b4f4b28cfd1d9d7d966b38':
Return display name in SipConnection.getCnapName().
|
| 6037a056ea0dda27a286ddcb527b323b58a1c7c7 |
|
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 |
|
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
|
| 99f92ac7a4e5907553cf737817bef06e429534a3 |
|
14-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am f5b7c855: am f5201ab7: Keep original phone number in SipConnection.
Merge commit 'f5b7c855fbb69e8614dab5ca15639228a2428784'
* commit 'f5b7c855fbb69e8614dab5ca15639228a2428784':
Keep original phone number in SipConnection.
|
| f5201ab71ff4d104265ab126e86afc6b81da8011 |
|
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
|
| a23935ab334b2dec47735608383608ca3093b4b2 |
|
12-Oct-2010 |
David Brown <dab@google.com> |
am bd6d5098: am d07833f5: Don\'t manually create CallerInfo objects from SipPhone
Merge commit 'bd6d5098a7995429713ae0ae43b3f666f7b9aed3'
* commit 'bd6d5098a7995429713ae0ae43b3f666f7b9aed3':
Don't manually create CallerInfo objects from SipPhone
|
| d07833f54b6e8e361b666ae16efa15fdf60159de |
|
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
|
| 026284745bb2f84e96fe132071f48a8cd4c1e715 |
|
08-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 88b4bd5f: am fc7e7766: Merge "CallManager/SipPhone: fix reject a ringing call" into gingerbread
Merge commit '88b4bd5f3e8d6e68541eec4a603b1df83291cf1f'
* commit '88b4bd5f3e8d6e68541eec4a603b1df83291cf1f':
CallManager/SipPhone: fix reject a ringing call
|
| f209cd70623f837026fb6c41e40a421291be62d0 |
|
07-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am a785a59c: am 718e0033: Merge "SIP: add SERVER_UNREACHABLE error code." into gingerbread
Merge commit 'a785a59c831256f274627f8f8eb77f9d54508916'
* commit 'a785a59c831256f274627f8f8eb77f9d54508916':
SIP: add SERVER_UNREACHABLE error code.
|
| 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
|
| aeba1bc0c333f145469fc17a10c0bbcebd6dc30b |
|
02-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Revert "Revert "resolved conflicts for merge of 8788d140 to master""
This reverts commit bdf11be97bd732e8891ae19342c937da6e659afa.
Fix a missing import from manual merge.
Change-Id: If373626f07250cbfe07e5c04cf02ad9ee5a0ab2a
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
| bdf11be97bd732e8891ae19342c937da6e659afa |
|
02-Oct-2010 |
Jason Chen <jasonchen@google.com> |
Revert "resolved conflicts for merge of 8788d140 to master"
This reverts commit 2874c3dec4f9ffd59b2be3de62c1148534396828, reversing
changes made to 7afbb30d636351334d101fd0caef391fa409230d.
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
| 2874c3dec4f9ffd59b2be3de62c1148534396828 |
|
01-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
resolved conflicts for merge of 8788d140 to master
Change-Id: I7eaf2b7fe968e8d4cf6c5a2a4e66b8584c1dc78c
|
| 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
|
| b6153a067ca6ec8affd782404525d6443e5f99b0 |
|
02-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 86a12d46: am 10e2120b: Merge "Add CallManager/Phone.setEchoSuppressionEnabled()." into gingerbread
Merge commit '86a12d465fca7d25d99a788abda33fde5154a739'
* commit '86a12d465fca7d25d99a788abda33fde5154a739':
Add CallManager/Phone.setEchoSuppressionEnabled().
|
| db9e87b98874bba9f26c8d0745639dfbe11195df |
|
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 49c13e51: am 0e412304: Catch exceptions in SipPhone.canTake().
Merge commit '49c13e512c675b27099783ccf2d20c9ef46c99db'
* commit '49c13e512c675b27099783ccf2d20c9ef46c99db':
Catch exceptions in SipPhone.canTake().
|
| bd407edc473778685c26d1c76a13671bf948eb83 |
|
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 2c457326: am 421c34c1: SipPhone: revise hangup() in SipCall and SipConnection.
Merge commit '2c4573263a97cc80f441824a75d6502ae04292a8'
* commit '2c4573263a97cc80f441824a75d6502ae04292a8':
SipPhone: revise hangup() in SipCall and SipConnection.
|
| 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
|
| 421c34c162098efe870574844a7ee49812bbb929 |
|
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
|
| 6c6eacda8066728537f2d8828e4c123f91ddfc27 |
|
28-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am f7e13400: am 624d5b4e: SIP: add DisconnectCause.SERVER_ERROR
Merge commit 'f7e13400b24bdc5461e640cfb2c9cb2bbb2b6236'
* commit 'f7e13400b24bdc5461e640cfb2c9cb2bbb2b6236':
SIP: add DisconnectCause.SERVER_ERROR
|
| 3c094de4a9d4793bba8f826a43ef84f509a743c4 |
|
28-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
am 4a64afd2: am 24547592: Fix the startAudio order for 3-way calls.
Merge commit '4a64afd2889cb6b6b2c94d3d7b24ebd3a2f10989'
* commit '4a64afd2889cb6b6b2c94d3d7b24ebd3a2f10989':
Fix the startAudio order for 3-way calls.
|
| c508e8468f54164e7c9729e3800e2559b803cef1 |
|
28-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Fix build.
Change-Id: I0c158e65eb47de64eb954a0e11c9843988cb8043
/frameworks/base/telephony/java/com/android/internal/telephony/sip/SipPhone.java
|
| e18f15e6fe9a7217de805e8a7ad84c01761910ec |
|
28-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 1738252a: am 32346522: Merge "Don\'t enter DISCONNECTING state when the call/connection is not alive" into gingerbread
Merge commit '1738252a596c71851cabf5835acb3584ad6b3191'
* commit '1738252a596c71851cabf5835acb3584ad6b3191':
Don't enter DISCONNECTING state when the call/connection is not alive
|
| 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
|
| 6cf8f64be3c079ef76d708a56b55a2b7ca6dbd2f |
|
27-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am c5027642: am 025a39af: SIP: misc fixes
Merge commit 'c5027642ff3909911d27e4abaa5e3abf1615b38d'
* commit 'c5027642ff3909911d27e4abaa5e3abf1615b38d':
SIP: misc fixes
|
| 031d8786824a385fa47750e5e8aa75f40d70cae9 |
|
27-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am fe2d279c: am 00a22064: SipService: handle cross-domain authentication error
Merge commit 'fe2d279c5ef571340f20d433badd9f68072299af'
* commit 'fe2d279c5ef571340f20d433badd9f68072299af':
SipService: handle cross-domain authentication error
|
| 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
|
| 5e18ad0c53faf88357c83bae66ab9d04c0388bb9 |
|
27-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
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.
|
| bd2294204e3edaede3fe81eb9b11c05c4fafe627 |
|
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
|
| 3011b6617232d1ef7c1687d1db9e780fce978d30 |
|
25-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am d6d83279: am 194bbcce: SIP: longer timeout for making call, shorter for cancelling
Merge commit 'd6d83279183db749de07bfdac79fe4180fc848d0'
* commit 'd6d83279183db749de07bfdac79fe4180fc848d0':
SIP: longer timeout for making call, shorter for cancelling
|
| 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
|
| 22a62d1342a973ed7d38bb7d1751a472365e2897 |
|
24-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
resolved conflicts for merge of ee813bae to master
Change-Id: I84ca961fb18b29313b5ec6041a79ca87a1a1cd68
|
| 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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| 6b15ecf2787dd0fa8e119785495112a84c30afe0 |
|
21-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
am 232bc085: am 708e4251: Merge "Revert the ANSWERING state." into gingerbread
Merge commit '232bc085fc901b6234d47c4ee4e3ee4fc88431e7'
* commit '232bc085fc901b6234d47c4ee4e3ee4fc88431e7':
Revert the ANSWERING state.
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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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| 5491c7a076bfee6db99ef96a966aadc4bb84b662 |
|
20-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 852e5354: am 8544560c: SipPhone: fix missing-call DisconnectCause feedback
Merge commit '852e5354f60a0131692c46f509c2e9901b0b6810'
* commit '852e5354f60a0131692c46f509c2e9901b0b6810':
SipPhone: fix missing-call DisconnectCause feedback
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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
|
| 2417132611936918cec91a85269a4e2e2752e9de |
|
20-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 37d1b249: am 97963794: SIP: convert enum to static final int.
Merge commit '37d1b2496ed4e636062d8bb451e723b975c80920'
* commit '37d1b2496ed4e636062d8bb451e723b975c80920':
SIP: convert enum to static final int.
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| 97963794af1e18674dd111e3ad344d90b16c922c |
|
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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| 0a56f342f9e75bc108f1e37944b61f754cf6168b |
|
17-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 960d409c: am 1d158357: SipPhone: do not append SIP domain to PSTN number
Merge commit '960d409c79aad3a9f78d930cdebedcc0fb34c30e'
* commit '960d409c79aad3a9f78d930cdebedcc0fb34c30e':
SipPhone: do not append SIP domain to PSTN number
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| 6c62609e8fd71e5c25b843a955caf4d41a3a5db7 |
|
17-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
resolved conflicts for merge of 394d1e4b to master
Change-Id: I2c3a0ff646e3024d422d809ce964bd188fc70bb7
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| 1d1583573d2099756bbbeef48d97c280edc393e0 |
|
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
|
| afa583e6557557577188c3e40146ac8d6f2aa7c7 |
|
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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| 6308514cbff740772534338ed641d3243cca52fe |
|
17-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 156edcc9: am 9404e633: Merge "Add timer to SIP session creation process." into gingerbread
Merge commit '156edcc9f64e010d6b6de97e9a77adfccee353eb'
* commit '156edcc9f64e010d6b6de97e9a77adfccee353eb':
Add timer to SIP session creation process.
|
| 9352cf1a4d46492fc48a20f7d825a9bcb6e8b365 |
|
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
|
| d6e15ab6aa387958222407b97cd21ae1d8b7965b |
|
16-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
am 170255b0: am d8f3d167: Add a new phone state ANSWERING.
Merge commit '170255b0f0482baf843b881a02d9361adcc33742'
* commit '170255b0f0482baf843b881a02d9361adcc33742':
Add a new phone state ANSWERING.
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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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| 03df1b5da60c5e2d3218937cd3978616f822e763 |
|
16-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 1e563f91: am 968735e5: Merge "Fixing the NPE in SipPhone bug id: http://b/2987816" into gingerbread
Merge commit '1e563f91183d95f5ad2461b7bca864e8f223ef71'
* commit '1e563f91183d95f5ad2461b7bca864e8f223ef71':
Fixing the NPE in SipPhone
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| 3e5246b2eb405b6ae304e7755ae5935a16e6ecc7 |
|
16-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 8a33e964: am 5306e0a8: Merge "SIP: add PEER_NOT_REACHABLE error feedback." into gingerbread
Merge commit '8a33e964c48d22469487c8ec1d951826b7e3e562'
* commit '8a33e964c48d22469487c8ec1d951826b7e3e562':
SIP: add PEER_NOT_REACHABLE error feedback.
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| b1c4a01985df67175e57f8fbea90706d2f579648 |
|
15-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am ae83faa3: am 13f6270e: SipAudioCall: use SipErrorCode instead of string in onError()
Merge commit 'ae83faa3eeb26b1983fb1b8b663eebfe1f1f61d5'
* commit 'ae83faa3eeb26b1983fb1b8b663eebfe1f1f61d5':
SipAudioCall: use SipErrorCode instead of string in onError()
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| 074663c7625e84caad1b6305c8b88968c28a4618 |
|
15-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am ca83c25d: am 4565933f: Merge "SipService: deliver connectivity change to all sessions." into gingerbread
Merge commit 'ca83c25decd0d4dc9c765a2b42424c7974bd496a'
* commit 'ca83c25decd0d4dc9c765a2b42424c7974bd496a':
SipService: deliver connectivity change to all sessions.
|
| 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
|
| 13f6270eb14b409709c936b828e2a2fd40e427c4 |
|
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
|
| 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
|
| 0b5a8bd57993f9a1c199c93fa3c9038fdece628d |
|
13-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am a5dce0c1: am 3d7606aa: SIP: enhance timeout and registration status feedback.
Merge commit 'a5dce0c15ac05724b4595d62d521a481c7e1f86a'
* commit 'a5dce0c15ac05724b4595d62d521a481c7e1f86a':
SIP: enhance timeout and registration status feedback.
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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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| e765994831da699bcead5597e3c5f0c7403d06cd |
|
13-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 38dc67f4: am 25b52a2f: SIP: remove dependency on javax.sip.SipException.
Merge commit '38dc67f49ec77d34c858777144323960c37d045e'
* commit '38dc67f49ec77d34c858777144323960c37d045e':
SIP: remove dependency on javax.sip.SipException.
|
| 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
|
| a97ccc02e18cd21c9cf1caaf63c4a680bf3c6f0a |
|
10-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 5f93c39c: am ca3c24db: Merge "SIP: add SipErrorCode for error feedback." into gingerbread
Merge commit '5f93c39cdb2f75dda805691987ccd4e570f6cb74'
* commit '5f93c39cdb2f75dda805691987ccd4e570f6cb74':
SIP: add SipErrorCode for error feedback.
|
| 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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| 577afc1c159b5f450ce16706747a3088299fd702 |
|
06-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
am b65fd472: am b6264a87: Fix the build.
Merge commit 'b65fd4726ce625958876ee0b68c622839f33a64c'
* commit 'b65fd4726ce625958876ee0b68c622839f33a64c':
Fix the build.
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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
|
| e2aded2d54400c950cc8450b34a641255bf8f3c1 |
|
06-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
am 2143ade8: am b12baad9: Add equals() funcation for SipPhone.
Merge commit '2143ade8030b8765e8744f5ff1ad90343f7fbfcd'
* commit '2143ade8030b8765e8744f5ff1ad90343f7fbfcd':
Add equals() funcation for SipPhone.
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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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| 63134b3aee2143c6df1ae9e2f8e4d516e26a0e2a |
|
24-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 1537637c: am 3294d44b: Add confcall management to SIP calls
Merge commit '1537637cc443a48e9afb8091b6f2ce329318f2d7'
* commit '1537637cc443a48e9afb8091b6f2ce329318f2d7':
Add confcall management to SIP calls
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| 3294d44b96f63f647fba3a03604eb028e28a42bc |
|
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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| 1c66bfb37edd9531c97ba2851338d521be73122c |
|
18-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
am dc7d7378: am 8eac20ea: SIP: implement conference call
Merge commit 'dc7d73783e0416cbfde6053a037ce32c8f35fbc4'
* commit 'dc7d73783e0416cbfde6053a037ce32c8f35fbc4':
SIP: implement conference call
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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
|
| a5d4503bc4697fa6e9a3a387dacfbe09050e400c |
|
11-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
am 2c9de2b9: am f57324cf: Add getSipUri() for identification.
Merge commit '2c9de2b9e5568c8918b36866d3cf20833a8c20e1'
* commit '2c9de2b9e5568c8918b36866d3cf20833a8c20e1':
Add getSipUri() for identification.
|
| 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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| 1601400b58790f06b53be87a47c7519d8f6f5a1d |
|
10-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
am f13542f2: am 259b4c86: Merge "SIP: clean up unused class and fields." into gingerbread
Merge commit 'f13542f2eba27d9b09ba6aa313f758e5518caa00'
* commit 'f13542f2eba27d9b09ba6aa313f758e5518caa00':
SIP: clean up unused class and fields.
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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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| 88e590fb370d80d863417aae9330c8c2218f3175 |
|
06-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Fix the SipPhone codes related to the ITelephone interface change.
Change-Id: Id8ab7a6e4feaac67cd09e8412af61b40c6e774b4
/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 |
|
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
|