summaryrefslogtreecommitdiff
path: root/voip/java/android/net/sip/SipManager.java
AgeCommit message (Collapse)Author
2013-01-28Delete frameworks/base/voip use voip-common from frameworks/opt/net/voipWink Saville
Change-Id: Ieaba759a0f69b45c4b8839cbed1fe757cdf190c5
2013-01-22Add debug and some cleanupWink Saville
Change-Id: I866676a1ec4a338dcfe089cbf0483e5e546ded85
2011-12-22docs: Add developer guide cross-references, Project ACRE, round 4Joe Fernandez
Change-Id: I1b43414aaec8ea217b39a0d780c80a25409d0991
2011-08-01Prevent NullPointerException cases while using SipService.Masahiko Endo
Some SipService methods may return null, in such cases like no Wi-Fi connection. Added minimum check to prevent NullPointerExceptions. Change-Id: Ia7fae57ee893f2564cbfdedb6dc614938ab60ff7 Signed-off-by: Masahiko Endo <masahiko.endo@gmail.com>
2010-12-21Check if VoIP API is supported in SipManager.Hung-ying Tyan
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
2010-11-03Throw proper exceptions in SipManagerHung-ying Tyan
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
2010-10-28docs: revise javadocs for sipScott Main
add a package description, revise class descriptions and edit some method docs Change-Id: Ice969a99c830349674c65d99e4b7a6f1d2f24a7e
2010-10-21Add permission requirements to SipAudioCall and SipManager javadoc.Hung-ying Tyan
Bug: 3116259 Change-Id: I00a033794e9d3e1c2d2ccfe4e612cd50003ec2ee
2010-10-20Remove ringtone API from SipAudioCall.Hung-ying Tyan
(watch out auto-merge conflict for SipAudioCall). Bug: 3113033, related CL: https://android-git/g/#change,75185 Change-Id: Ib48d3b990e229e0b341e47e10e76934e1a50d10f
2010-10-18Uncomment SIP/VOIP feature check in SipManager.Hung-ying Tyan
http://b/issue?id=2971947 Change-Id: I3afa8eb03c4e347b382213dd388354365f766b2f
2010-10-08Unhide SIP API.Hung-ying Tyan
Change-Id: I09468e3149a242a3b1e085ad220eb74f84ac6c68
2010-10-07Make SipService broadcast SIP_SERVICE_UP when it's up.Hung-ying Tyan
http://b/issue?id=3062010 Change-Id: I13419fa3a8fdfba1977260f703e4dcaa42a6606c
2010-10-05SipService: supply PendingIntent when open a profile.Hung-ying Tyan
The SipService used to take an action string and broadcasts an intent with that action string when an incoming call is received. The design is not safe (as the intent may be sniffed) and inflexible (can only received by BroadcastReceiver). Now we use PendingIntent to fix all these. Companion CL: https://android-git.corp.google.com/g/#change,71800 Change-Id: Id12e5c1cf9321edafb171494932cd936eae10b6e
2010-10-04SIP: minor fixes.Hung-ying Tyan
+ Log error instead of crashing app process in SipManager's ListenerRelay. + Terminate dialog and transaction in SipSessionGroup.reset(). + Remove redundant reset() in SipSessionGroup. Change-Id: Ifbf29d2c9607ffe1a1a50b0c131ee3a4e81a0d0e
2010-09-30Add uri field to SipManager.ListenerRelayHung-ying Tyan
in case mSession is not available. Change-Id: Ifee2c129e48aa1177f648f176413ab6aa5606770
2010-09-30SIP: misc fixes.Hung-ying Tyan
+ Fix keepalive timer event leak due to the race between stopping timer and the async'ed timeout handler + SipSessionImpl: set state before handling an event to ensure we get correct state when some error occurs during handling the event. + Fix potential NPE in SipManager.ListenerRelay.getUri(). Change-Id: I021ee34f83059fd4fbb64b30bea427a5462aa51b
2010-09-24SipAudioCall: remove SipManager dependency.Hung-ying Tyan
Change-Id: I2dc8bf427e52f64529ee0e0261362b975a8917c6
2010-09-24Refactoring SIP classes to get ready for API review.Hung-ying Tyan
+ 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
2010-09-20SIP: convert enum to static final int.Hung-ying Tyan
Converts SipErrorCode and SipSessionState. Change-Id: Iee3a465649ea89d395b2336bbd673c25113e5f93
2010-09-20SIP: add config flag for wifi-only configuration.Hung-ying Tyan
http://b/issue?id=2994029 Change-Id: I328da9b0f8b70d660dbcefffdac8250341792101
2010-09-17Add timer to SIP session creation process.Hung-ying Tyan
+ 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
2010-09-16Fix links in SIP API javadoc.Hung-ying Tyan
Change-Id: I839280fe18502bb576f6e9c9a7948077c02fa570
2010-09-14SIP: remove dependency on javax.sipHung-ying Tyan
and change errorCodeString to errorCode in SipRegistrationListener.onRegistrationFailed(). Change-Id: Id9618f5a4b0effaed04f8b0dc60347499d9e4501
2010-09-13SIP: remove dependency on javax.sip.SipException.Hung-ying Tyan
Change-Id: I77d289bef1b5e7f1ec0c0408d0bbf96c21085cd7
2010-09-03SipManager: always return true for SIP API and VOIP support query.Hung-ying Tyan
Change-Id: I397a804e0aa598aee77a8ce28ada1b11e10fbaea http://b/issue?id=2972054
2010-09-02SipService: reduce the usage of javax.sdp.*.Chia-chi Yeh
After this change, SipAudioCallImpl is the only place still using it. Change-Id: I5693bffa54f9e19cbfa70b45dfcf40fba04dedbb
2010-09-02Add software features for SIP and VOIPHung-ying Tyan
and block SipService creation and SIP API if the feature is not available. Change-Id: Icf780af1ac20dda4d8180cea3e5b20e21a8350bc
2010-08-10SIP: clean up unused class and fields.Hung-ying Tyan
Change-Id: I79ed7fb324fea9a52946340055b5ea1d389a926a
2010-08-05Revert "Move SIP telephony related codes to framework."Chung-yih Wang
This reverts commit b631dcf3eb449ddec756bea330f4e70b996ffb9e.
2010-08-05Move SIP telephony related codes to framework.Chung-yih Wang
+ hardcode the sip service for build dependency. Change-Id: Ib0e9717c9b87eb6e06ffa3a7b01ae31184de61bb
2010-08-05Move the sip related codes to framework.Chung-yih Wang
Change-Id: Ib81dadc39b73325c8438f078c7251857a83834fe
generated by cgit v1.2.3 (git 2.25.1) at 2025-10-03 19:47:04 +0000