| 3aef8e1d1b2f0b87d470bcccf37ba4ebb6560c45 |
|
20-Dec-2011 |
Joe Fernandez <joefernandez@google.com> |
docs: Add developer guide cross-references, Project ACRE, round 4
Change-Id: I1b43414aaec8ea217b39a0d780c80a25409d0991
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|
| 307f15faafa5a38d9b3b314df22778cd11685d7b |
|
12-Jul-2011 |
repo sync <cywang@google.com> |
Add REFER handling.
Handle REFER requests including REFER with Replaces header.
bug:4958680
Change-Id: I96df95097b78bed67ab8abd309a1e57a45c6bc2f
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|
| 2093561a58e602450f6e4f2aae4831edd1b840f4 |
|
28-Jun-2011 |
repo sync <cywang@google.com> |
Support INVITE w/o SDP.
bug:3326873
Change-Id: Ie29d2c61b237fee2d8637f4ba3d293a22469cced
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|
| 1aceda35cc607856ec2e960e0c6cfc6aea87ab8e |
|
23-Jun-2011 |
repo sync <cywang@google.com> |
Support Invite w/ Replaces request.
bug:3326870
Change-Id: Idbfbe7e3cc6ba83874d42bfb7d149866f454e70a
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|
| 89bc1fe73efb73b43758d41c9ff9f2f4902dd019 |
|
25-Feb-2011 |
Chung-yih Wang <cywang@google.com> |
Activate the wifi high perf. for sip calls.
bug:3487791
Change-Id: I7d8d146f8542cd7df387547c7ce3d5ded27f8e97
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|
| 4bf82df2f069b5a788689064bf8d3f6b612587d4 |
|
06-Jan-2011 |
Chia-chi Yeh <chiachi@android.com> |
Do not set back to AudioManager.MODE_NORMAL in SipAudioCall.
Change-Id: I8f68e01e5f8c73bb8afd44312cbfadb55aab4330
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|
| 33808c6d2448bbc944905819c213f2debf18af5a |
|
22-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
am aec9a33f: am e0bd2688: Merge "Check if VoIP API is supported in SipManager." into gingerbread
* commit 'aec9a33f1cfc7c32690bc8e24aefaeb137ab9859':
Check if VoIP API is supported in SipManager.
|
| 5bd3782f244212cd8ef51bf9f3578869b08b4e18 |
|
20-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
Check if VoIP API is supported in SipManager.
This is to make SipManager.isVoipSupported() effective.
Also add NPE check now that we may return null SipAudioCall when VOIP is not
supported.
Bug: 3251016
Change-Id: Icd551123499f55eef190743b90980922893c4a13
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|
| 2aef9a1e847a7612549d9a0280cde6489e540f6b |
|
03-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 7da1ffc9: am e2abd103: Merge "Set AudioGroup mode according to audio settings" into gingerbread
* commit '7da1ffc9d2a51ef6120389a06351fd770ab45f46':
Set AudioGroup mode according to audio settings
|
| fa81463e88d15859b557be6fef5982b049b92ab8 |
|
25-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Set AudioGroup mode according to audio settings
Set AudioGroup mode according to holding, mute and speaker phone settings.
Bug: 3119690
Change-Id: I02803ae105409b7f8482e6c2ef3e67623bd54e03
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|
| e5bc8f617b48ab237bec22dd4572e678642f25eb |
|
29-Oct-2010 |
Scott Main <smain@google.com> |
am 9a8df805: am 1112632a: Merge "docs: revise javadocs for sip add a package description, revise class descriptions and edit some method docs" into gingerbread
* commit '9a8df8054b2e38a27d8e8e6b17365979218f0e3f':
docs: revise javadocs for sip add a package description, revise class descriptions and edit some method docs
|
| 02b1d685cc287d7c53141872b3d80be4ee5dd59e |
|
22-Oct-2010 |
Scott Main <smain@google.com> |
docs: revise javadocs for sip
add a package description, revise class descriptions and edit some method docs
Change-Id: Ice969a99c830349674c65d99e4b7a6f1d2f24a7e
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|
| 6d848f759e901264935ed7ba1094d865e3b2c16b |
|
21-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am bdc15d8b: am 4056ab97: Merge "Add permission requirements to SipAudioCall and SipManager javadoc." into gingerbread
Merge commit 'bdc15d8b43e7763bf72780d0f683b223b8cf6625'
* commit 'bdc15d8b43e7763bf72780d0f683b223b8cf6625':
Add permission requirements to SipAudioCall and SipManager javadoc.
|
| 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.
|
| e87b644402642bad7147f915849bfa0eadaea446 |
|
18-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Add permission requirements to SipAudioCall and SipManager javadoc.
Bug: 3116259
Change-Id: I00a033794e9d3e1c2d2ccfe4e612cd50003ec2ee
/frameworks/base/voip/java/android/net/sip/SipAudioCall.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/voip/java/android/net/sip/SipAudioCall.java
|
| 3d59480dc201c893c6da5c3934b14a2d95a1bef9 |
|
10-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am ea445758: am 08faac3c: Unhide SIP API.
Merge commit 'ea445758efba6b728d5e597402e9d9538f3ef451'
* commit 'ea445758efba6b728d5e597402e9d9538f3ef451':
Unhide SIP API.
|
| 08faac3c26e12863858e1534985dd950193f755f |
|
15-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Unhide SIP API.
Change-Id: I09468e3149a242a3b1e085ad220eb74f84ac6c68
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|
| 5a474a2bb8bc23fcc8d05e8b9ec3f4306dd63db1 |
|
27-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 44669d31: am fd144d76: Merge "SipAudioCall: remove SipManager dependency." into gingerbread
Merge commit '44669d31d1d5b094d7b7d3e393281440ea0c9536'
* commit '44669d31d1d5b094d7b7d3e393281440ea0c9536':
SipAudioCall: remove SipManager dependency.
|
| 3a4197e642e9c70f1fe00c2cba30f0f957d36bfc |
|
24-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipAudioCall: remove SipManager dependency.
Change-Id: I2dc8bf427e52f64529ee0e0261362b975a8917c6
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|
| fb0264096e08aeeb350c9a2762b34d14361ba38e |
|
24-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
fix build
Change-Id: Iff05b5ea7f535f532eec2af1edf78fdf8acfa21c
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|
| 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/voip/java/android/net/sip/SipAudioCall.java
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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/voip/java/android/net/sip/SipAudioCall.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/voip/java/android/net/sip/SipAudioCall.java
|
| 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/voip/java/android/net/sip/SipAudioCall.java
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| 286bb5a00bdb9f0cb0815aef441ec72f231c84ea |
|
14-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Fix links in SIP API javadoc.
Change-Id: I839280fe18502bb576f6e9c9a7948077c02fa570
/frameworks/base/voip/java/android/net/sip/SipAudioCall.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/voip/java/android/net/sip/SipAudioCall.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/voip/java/android/net/sip/SipAudioCall.java
|
| 903e1031605d715e904811b0dd06cc6a518f0048 |
|
09-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: add SipErrorCode for error feedback.
Change-Id: I8b071d4933479b780a403d0bfa30511f4c23ca8f
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
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| 95b15c35608fe3ea679c8a478c6cbd841623371e |
|
02-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
SipService: reduce the usage of javax.sdp.*.
After this change, SipAudioCallImpl is the only place still using it.
Change-Id: I5693bffa54f9e19cbfa70b45dfcf40fba04dedbb
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
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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/voip/java/android/net/sip/SipAudioCall.java
|
| cfd15dd3c8554cbbcb5822a0fdf6ca31d6b28acf |
|
16-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Fix the IN_CALL mode issue.
If the sip call is on-holding, we should not set the audio to
MODE_NORMAL, or it will affect the audio if there is an active pstn
call.
Change-Id: If1bcba952617bf8427bc9e2d64d483ba1ee37370
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|
| 363c2ab82cca4f095e9e0c8465e28f6d27a24bf8 |
|
05-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Move the sip related codes to framework.
Change-Id: Ib81dadc39b73325c8438f078c7251857a83834fe
/frameworks/base/voip/java/android/net/sip/SipAudioCall.java
|