| 7314532349e402315af9b8f664432dd18292421f |
29-Aug-2012 |
Johan Redestig <johan.redestig@sonymobile.com> |
Make SimpleSessionDescription locale safe
Explicitly use Locale.US in SimpleSessionDescription to avoid
unexpected results in some locales.
Change-Id: Idb4a36a9e332d302e1b9b940355917c0f738e076
et/sip/SimpleSessionDescription.java
|
| 9be0105fbc56eb1b1813bb7c5fe258a144867a43 |
22-Jun-2012 |
Scott Main <smain@google.com> |
docs: fix several links
Change-Id: I89d9fd64dc22c90680bb05415cc966c255165af9
et/sip/package.html
|
| e66950506c473e660f2e5762d7a71e13808be387 |
30-Mar-2012 |
Chia-chi Yeh <chiachi@android.com> |
RTP: refactor a little bit and fix few minor bugs.
Change-Id: I063644507f26996ded462972afcb550a4528dac8
et/rtp/AudioGroup.java
et/rtp/AudioStream.java
et/rtp/RtpStream.java
|
| fb982db41060a2914cddb43200f3ee53627f8762 |
28-Mar-2012 |
Chia-chi Yeh <chiachi@android.com> |
RTP: add a null-check in AudioStream.setDtmfType().
Change-Id: I52cbdea48affae3747942940451f4fd5ca47030f
et/rtp/AudioStream.java
|
| 3aef8e1d1b2f0b87d470bcccf37ba4ebb6560c45 |
20-Dec-2011 |
Joe Fernandez <joefernandez@google.com> |
docs: Add developer guide cross-references, Project ACRE, round 4
Change-Id: I1b43414aaec8ea217b39a0d780c80a25409d0991
et/sip/SipAudioCall.java
et/sip/SipManager.java
et/sip/SipProfile.java
et/sip/package.html
|
| dc5bbe965f7a66238c3f9a6694f4410b3d52af3b |
14-Aug-2011 |
Hung-ying Tyan <tyanh@google.com> |
Handle SIP authentication response for BYE.
Bug: 5159669
Change-Id: I029684334500d4d0db176783084c9b7d1db87e40
et/sip/SipSession.java
|
| 53ad2c7fe212a08ae05fb4d7f27d42f9a0a4b912 |
02-Aug-2011 |
Conley Owens <cco3@android.com> |
am 0793586b: am f8c1f129: am e1d27154: am f87743e7: Merge "Prevent NullPointerException cases while using SipService."
* commit '0793586bf8f4dce71d0b4d7ff2f212129b3f76fe':
Prevent NullPointerException cases while using SipService.
|
| 0793586bf8f4dce71d0b4d7ff2f212129b3f76fe |
02-Aug-2011 |
Conley Owens <cco3@android.com> |
am f8c1f129: am e1d27154: am f87743e7: Merge "Prevent NullPointerException cases while using SipService."
* commit 'f8c1f1298ac3ede518c8d29eeb6719746c6afaf0':
Prevent NullPointerException cases while using SipService.
|
| 25ccbb97ffd3298caede635f29445073e845cfc3 |
28-Jul-2011 |
Masahiko Endo <masahiko.endo@gmail.com> |
Prevent NullPointerException cases while using SipService.
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>
et/sip/SipManager.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
et/sip/SipAudioCall.java
|
| 2093561a58e602450f6e4f2aae4831edd1b840f4 |
28-Jun-2011 |
repo sync <cywang@google.com> |
Support INVITE w/o SDP.
bug:3326873
Change-Id: Ie29d2c61b237fee2d8637f4ba3d293a22469cced
et/sip/SipAudioCall.java
|
| 1aceda35cc607856ec2e960e0c6cfc6aea87ab8e |
23-Jun-2011 |
repo sync <cywang@google.com> |
Support Invite w/ Replaces request.
bug:3326870
Change-Id: Idbfbe7e3cc6ba83874d42bfb7d149866f454e70a
et/sip/ISipSessionListener.aidl
et/sip/SipAudioCall.java
et/sip/SipSession.java
et/sip/SipSessionAdapter.java
|
| de9acb76d9ea398d0ba4c5e62df554f5696eaa99 |
06-May-2011 |
Scott Main <smain@google.com> |
docs: add package description for RTP
Change-Id: I02c181a48101be288fb4aabf497f573f00038f90
et/rtp/package.html
|
| 6defd2d47e81b206d76430266120294a40592b27 |
03-Mar-2011 |
Chia-chi Yeh <chiachi@android.com> |
NEW_API: Unhide RTP APIs.
This change unhides RTP related classes including AudioCodec,
AudioGroup, AudioStream, and RtpStream. This allows developers
to control audio streams directly and also makes conference
calls possible with the combination of the public SIP APIs.
Change-Id: Idfd4edf65a1cbf3245ec2786fbc03b06438b0fb3
et/rtp/AudioCodec.java
et/rtp/AudioGroup.java
et/rtp/AudioStream.java
et/rtp/RtpStream.java
|
| c52f5b2ec5e13ab3d9ab016e6cab757d4ecb45c7 |
03-Mar-2011 |
Chia-chi Yeh <chiachi@android.com> |
RTP: update javadocs.
Change-Id: If600df0eb1e6135aed9f3b2eacfb6bc9ed5d78ff
et/rtp/AudioGroup.java
et/rtp/AudioStream.java
|
| 89bc1fe73efb73b43758d41c9ff9f2f4902dd019 |
25-Feb-2011 |
Chung-yih Wang <cywang@google.com> |
Activate the wifi high perf. for sip calls.
bug:3487791
Change-Id: I7d8d146f8542cd7df387547c7ce3d5ded27f8e97
et/sip/SipAudioCall.java
|
| 9e25df44631e3c7881a6816cf26f34ea24055c72 |
10-Feb-2011 |
Chung-yih Wang <cywang@google.com> |
Make SIP AuthName APIs public.
bug:3326867
Change-Id: I766e6e28f6ad3e84de2c9e24850d472ad00271cc
et/sip/SipProfile.java
|
| 0f7de88cb9eef781117fa2f2b69ba2698237637e |
06-Jan-2011 |
Chung-yih Wang <cywang@google.com> |
Merge "Add auth. username in SipProfile." from gingerbread.
bug:3326867
Change-Id: Ic67dd7d4858f28224e4f01ad8b65bcd3a3c15f10
et/sip/SipProfile.java
|
| f268a2f8488b6b111126a7043a5f1f559a566fa7 |
06-Jan-2011 |
Chung-yih Wang <cywang@google.com> |
Add auth. username in SipProfile.
bug:3326867
Change-Id: I2a62c75fb3f5e9c6ec2e00b29396e93b0c183d9b
et/sip/SipProfile.java
|
| 4bf82df2f069b5a788689064bf8d3f6b612587d4 |
06-Jan-2011 |
Chia-chi Yeh <chiachi@android.com> |
Do not set back to AudioManager.MODE_NORMAL in SipAudioCall.
Change-Id: I8f68e01e5f8c73bb8afd44312cbfadb55aab4330
et/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
et/sip/SipAudioCall.java
et/sip/SipManager.java
|
| 58ee2acba8953814cc4bf65d2f28f7dd498b5779 |
16-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
Check port in create peer's SIP profile.
SipURI returns port -1 when port is not present in the URI.
Don't call SipProfile.Builder.setPort() when that happens.
Bug: 3291248
Change-Id: I8e608cbc56ea82862df55fdba885f6a864db83ab
et/sip/SipProfile.java
|
| eecf4a6f11129461088d620afadb6014edab3086 |
16-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
Check port in create peer's SIP profile.
SipURI returns port -1 when port is not present in the URI.
Don't call SipProfile.Builder.setPort() when that happens.
Change-Id: Ic5fe7301195705a77010038cae20d6629b33135e
et/sip/SipProfile.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
et/sip/SipProfile.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
|
| 4c7cc83827458945fe7a1f4bd2bfe0629f0d30ae |
01-Dec-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: Prepare to unhide the APIs."
|
| 53aa6ef70d8692277f9403f94d43918ad9712dd0 |
30-Nov-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Prepare to unhide the APIs.
Polish things a little bit.
Change-Id: I2c3cea8b34b9c858879bc722ea1f38082ba22b8d
et/rtp/AudioGroup.java
et/rtp/AudioStream.java
et/rtp/RtpStream.java
|
| 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
|
| 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
et/sip/SipAudioCall.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
et/sip/SipManager.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
et/sip/SipAudioCall.java
et/sip/SipErrorCode.java
et/sip/SipException.java
et/sip/SipManager.java
et/sip/SipProfile.java
et/sip/SipRegistrationListener.java
et/sip/SipSession.java
et/sip/package.html
|
| 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
et/sip/SipAudioCall.java
et/sip/SipManager.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
et/sip/SipAudioCall.java
et/sip/SipManager.java
|
| 78c206c75095b4a9092aabed8e79702ae0c5dc86 |
19-Oct-2010 |
John Huang <jsh@google.com> |
am 085996c4: am 45bd8303: Merge "Uncomment SIP/VOIP feature check in SipManager." into gingerbread
Merge commit '085996c411b4d3878dfd97c59bfc4a17da08959b'
* commit '085996c411b4d3878dfd97c59bfc4a17da08959b':
Uncomment SIP/VOIP feature check in SipManager.
|
| a0cdfbf5b74a92611789b7ec08a84274b9011021 |
18-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Uncomment SIP/VOIP feature check in SipManager.
http://b/issue?id=2971947
Change-Id: I3afa8eb03c4e347b382213dd388354365f766b2f
et/sip/SipManager.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.
|
| c7e4b2d5bb9c9ab2d4a8efa0fd07be9d83987d36 |
09-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 841d6ff9: am 62ec9834: Merge "Make SipService broadcast SIP_SERVICE_UP when it\'s up." into gingerbread
Merge commit '841d6ff9e05daccbc60daa1618a27e9db3a4fb32'
* commit '841d6ff9e05daccbc60daa1618a27e9db3a4fb32':
Make SipService broadcast SIP_SERVICE_UP when it's up.
|
| 08faac3c26e12863858e1534985dd950193f755f |
15-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Unhide SIP API.
Change-Id: I09468e3149a242a3b1e085ad220eb74f84ac6c68
et/sip/SipAudioCall.java
et/sip/SipErrorCode.java
et/sip/SipException.java
et/sip/SipManager.java
et/sip/SipProfile.java
et/sip/SipRegistrationListener.java
et/sip/SipSession.java
|
| 9db99a4dc10ac0d5d3751f03ea51c0fed217d2f8 |
07-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Make SipService broadcast SIP_SERVICE_UP when it's up.
http://b/issue?id=3062010
Change-Id: I13419fa3a8fdfba1977260f703e4dcaa42a6606c
et/sip/SipManager.java
|
| 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.
|
| 828c89ba8e76224bc61702fa3b9c093300825ba0 |
07-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 3cb2d3be: am 1862af57: Merge "SipService: supply PendingIntent when open a profile." into gingerbread
Merge commit '3cb2d3be6cb501c77c7a5765d954363125857cca'
* commit '3cb2d3be6cb501c77c7a5765d954363125857cca':
SipService: supply PendingIntent when open a profile.
|
| 718e0033e69fa7d1db12242324ab9098ac430bf5 |
05-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "SIP: add SERVER_UNREACHABLE error code." into gingerbread
|
| 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
et/sip/SipErrorCode.java
|
| 323d3671ac813df8dd173f3f4d6cb681ee29f740 |
05-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipService: supply PendingIntent when open a profile.
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
et/sip/ISipService.aidl
et/sip/SipManager.java
|
| 51d2adab83837425dae8062b7ff2a5bd1e732dd9 |
04-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 1f34ffd7: am 5cab38ba: Merge "SIP: minor fixes." into gingerbread
Merge commit '1f34ffd7e36de5d1a12d4a3901c3ac4e4e56cb99'
* commit '1f34ffd7e36de5d1a12d4a3901c3ac4e4e56cb99':
SIP: minor fixes.
|
| 9ea96c6cade1f25d4d77dcbd24854df431548b36 |
03-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: minor fixes.
+ 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
et/sip/SipManager.java
|
| e0ed9dbcb8f3b67f66a1b2a1df264e3aee0bb81c |
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am c79e74ec: am d29e0754: Merge "Add uri field to SipManager.ListenerRelay" into gingerbread
Merge commit 'c79e74ec1d30f95de89568ee645a9b1577ae73b3'
* commit 'c79e74ec1d30f95de89568ee645a9b1577ae73b3':
Add uri field to SipManager.ListenerRelay
|
| a77c9541d008cfffed71cb8e3a9382001cf7fe9c |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
am cbee6229: am 0a537b78: Merge "RTP: Enable AMR codec." into gingerbread
Merge commit 'cbee622954de5e9e0c07557f8ec9aaa741110043'
* commit 'cbee622954de5e9e0c07557f8ec9aaa741110043':
RTP: Enable AMR codec.
|
| d161479237010cb2b7bc8dab0fbbce2cf0170ecf |
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 947d2abd: am 2365b78e: Merge "SIP: misc fixes." into gingerbread
Merge commit '947d2abd82ef68c661fc29fd5167e4c0ba749f71'
* commit '947d2abd82ef68c661fc29fd5167e4c0ba749f71':
SIP: misc fixes.
|
| 5a7c6d298e9f8963e3b82f84da15f16a4a83f8ff |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
am 1c2eab2d: am 955ab37c: Merge "RTP: Enable GSM-EFR codec." into gingerbread
Merge commit '1c2eab2d86faa9c647a9893f761a50cfa28d9d57'
* commit '1c2eab2d86faa9c647a9893f761a50cfa28d9d57':
RTP: Enable GSM-EFR codec.
|
| 9e1d308e993d451882456e44cfaacae63df7a496 |
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Add uri field to SipManager.ListenerRelay
in case mSession is not available.
Change-Id: Ifee2c129e48aa1177f648f176413ab6aa5606770
et/sip/SipManager.java
|
| 0a537b78d3fb4db86411d745b2696459d6b98ef6 |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: Enable AMR codec." into gingerbread
|
| 2365b78e64feaa9527efb15bf4ac207a837f2b45 |
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "SIP: misc fixes." into gingerbread
|
| f88fc1fa907f720df4a3e915509e688e9e4cf1f8 |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Enable AMR codec.
Change-Id: I49e6bdc1b67306b44173f2f346f8372a50264870
et/rtp/AudioCodec.java
|
| fb3a98b1d8d0ad040980d509c4c5341928b9460b |
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: misc fixes.
+ 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
et/sip/SipManager.java
|
| f4ae94229d736c7dbd3c5c36d484213d51545702 |
29-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Enable GSM-EFR codec.
Change-Id: I9d84009e4557a0a82c1f9d7d543922741be97c77
et/rtp/AudioCodec.java
|
| dcf2be6cf660269c77f51ff0e0f336726d1625c6 |
29-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
am ebfe5632: am e006e4d2: Merge changes Iae1913fb,I38dbefef into gingerbread
Merge commit 'ebfe5632db275a89b49ab828064ba90db59702cf'
* commit 'ebfe5632db275a89b49ab828064ba90db59702cf':
RTP: Enable GSM codec.
RTP: Refactor out G711 codecs into another file.
|
| a6f950c9682ffffc00ca976aafeeedf391718b1d |
29-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Enable GSM codec.
Change-Id: Iae1913fb0643f1c66b5d16f24d51924d363e5ef5
et/rtp/AudioCodec.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.
|
| 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
|
| fd144d7667d9d050b7fb158276ae4623d4ea83b8 |
27-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "SipAudioCall: remove SipManager dependency." into gingerbread
|
| 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
et/sip/SipErrorCode.java
|
| 3a4197e642e9c70f1fe00c2cba30f0f957d36bfc |
24-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipAudioCall: remove SipManager dependency.
Change-Id: I2dc8bf427e52f64529ee0e0261362b975a8917c6
et/sip/SipAudioCall.java
et/sip/SipManager.java
|
| a97c5f7779bbd53896b5312c9dd04c505511781d |
24-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "fix build"
|
| fb0264096e08aeeb350c9a2762b34d14361ba38e |
24-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
fix build
Change-Id: Iff05b5ea7f535f532eec2af1edf78fdf8acfa21c
et/sip/SipAudioCall.java
|
| 658bec956785e074edc4f6c9fe739c366e37be33 |
23-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
SDP: remove dead code.
Change-Id: I2a5764a2b9cabc54b0ac18666e494c1cb39c4e9b
et/sip/SdpSessionDescription.java
et/sip/SessionDescription.aidl
et/sip/SessionDescription.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
et/sip/SipAudioCall.java
et/sip/SipAudioCallImpl.java
et/sip/SipManager.java
et/sip/SipProfile.java
et/sip/SipSession.java
et/sip/SipSessionState.java
|
| 0b7d6de1559a4a78af76ab501e0a15afc396c2b9 |
23-Sep-2010 |
repo sync <chiachi@android.com> |
Fix the build.
Change-Id: I82210cb2d41f532583f83ea17e6f2d8d49280a30
et/sip/SimpleSessionDescription.java
|
| 84f7f6ba3913a4ad8546d425197a6d64593b91cf |
23-Sep-2010 |
repo sync <chiachi@android.com> |
SIP: Make SipAudioCallImpl use SimpleSessionDescription instead of javax.sdp.
Change-Id: I7efff4f29ca84c3e7c17ef066b7186b514a777b2
et/sip/SipAudioCallImpl.java
|
| e6c0c109588771a97aba51d06fdf73557b06dfd3 |
20-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
SDP: Add a simple class to help manipulate session descriptions.
Change-Id: I1631ee20e8b4a9ad8e2184356b5d13de66e03db1
et/sip/SimpleSessionDescription.java
|
| 37adc522f6bc074a688ffbef420a8627ef9a4b5b |
21-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Add two getters to retrieve the current configuration from AudioStream.
Change-Id: Iff588130653242f6ddd6a6b663df775ecb276768
et/rtp/AudioStream.java
|
| 32e106b7bdd57c82ee67705871f6116d92bce79b |
16-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Extend codec capability and update the APIs.
Change-Id: I37ba9d83c2de3c5dae2bfc1b7513df7f6fee3c5c
et/rtp/AudioCodec.java
et/rtp/AudioGroup.java
et/rtp/AudioStream.java
et/rtp/RtpStream.java
|
| 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
et/sip/SipAudioCallImpl.java
|
| 97963794af1e18674dd111e3ad344d90b16c922c |
17-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: convert enum to static final int.
Converts SipErrorCode and SipSessionState.
Change-Id: Iee3a465649ea89d395b2336bbd673c25113e5f93
et/sip/ISipSession.aidl
et/sip/ISipSessionListener.aidl
et/sip/SipAudioCall.java
et/sip/SipAudioCallImpl.java
et/sip/SipErrorCode.java
et/sip/SipManager.java
et/sip/SipRegistrationListener.java
et/sip/SipSessionAdapter.java
et/sip/SipSessionState.java
|
| c4b87477c076d61062950becc132b7483e3fb198 |
19-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: add config flag for wifi-only configuration.
http://b/issue?id=2994029
Change-Id: I328da9b0f8b70d660dbcefffdac8250341792101
et/sip/SipManager.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
et/sip/SipAudioCall.java
et/sip/SipAudioCallImpl.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
et/sip/ISipSession.aidl
et/sip/SipAudioCall.java
et/sip/SipAudioCallImpl.java
et/sip/SipManager.java
|
| 286bb5a00bdb9f0cb0815aef441ec72f231c84ea |
14-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Fix links in SIP API javadoc.
Change-Id: I839280fe18502bb576f6e9c9a7948077c02fa570
et/sip/ISipSession.aidl
et/sip/SipAudioCall.java
et/sip/SipErrorCode.java
et/sip/SipException.java
et/sip/SipManager.java
et/sip/SipProfile.java
et/sip/SipRegistrationListener.java
|
| 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
et/sip/SipErrorCode.java
|
| 12bec5ddf58ad3a69728810480e6194c806567d6 |
14-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipService: ignore connect event for non-active networks.
+ sanity check and remove redundant code.
Change-Id: I4d3e226851ad7fc4d88ddcd0a5c58f7e33b6c14a
et/sip/SipAudioCallImpl.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
et/sip/SipAudioCall.java
et/sip/SipAudioCallImpl.java
|
| 99bf4e45c4566172189735b34b368b76660ca57a |
14-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: remove dependency on javax.sip
and change errorCodeString to errorCode in
SipRegistrationListener.onRegistrationFailed().
Change-Id: Id9618f5a4b0effaed04f8b0dc60347499d9e4501
et/sip/ISipSession.aidl
et/sip/SipManager.java
et/sip/SipProfile.java
et/sip/SipRegistrationListener.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
et/sip/SipErrorCode.java
|
| 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
et/sip/SipErrorCode.java
|
| 25b52a2f97df112c2836972d0b6d9a4c7a9c4a4e |
13-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: remove dependency on javax.sip.SipException.
Change-Id: I77d289bef1b5e7f1ec0c0408d0bbf96c21085cd7
et/sip/SipAudioCall.java
et/sip/SipAudioCallImpl.java
et/sip/SipException.java
et/sip/SipManager.java
|
| 903e1031605d715e904811b0dd06cc6a518f0048 |
09-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: add SipErrorCode for error feedback.
Change-Id: I8b071d4933479b780a403d0bfa30511f4c23ca8f
et/sip/ISipSessionListener.aidl
et/sip/SipAudioCall.java
et/sip/SipAudioCallImpl.java
et/sip/SipErrorCode.java
et/sip/SipRegistrationListener.java
et/sip/SipSessionAdapter.java
|
| 643fce978152c6c5ded316a8c9de6531b7d4cee7 |
03-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipManager: always return true for SIP API and VOIP support query.
Change-Id: I397a804e0aa598aee77a8ce28ada1b11e10fbaea
http://b/issue?id=2972054
et/sip/SipManager.java
|
| dc296b0d4bd6fef8764c10fb4cd59c85bc5186f6 |
02-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "SipService: reduce the usage of javax.sdp.*." into gingerbread
|
| 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
et/sip/ISipSession.aidl
et/sip/ISipSessionListener.aidl
et/sip/SdpSessionDescription.java
et/sip/SipAudioCall.java
et/sip/SipAudioCallImpl.java
et/sip/SipManager.java
et/sip/SipSessionAdapter.java
|
| 60264b306453a3043442719b970f2edb3f46f51b |
01-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipProfile: remove outgoingCallAllowed flag.
Change-Id: I37a215bafce57adf6911c81fd38db324bac686ec
et/sip/SipProfile.java
|
| 3424c02e6b931a8bbd651ae75217bebd008b2605 |
27-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
Add software features for SIP and VOIP
and block SipService creation and SIP API if the feature is not available.
Change-Id: Icf780af1ac20dda4d8180cea3e5b20e21a8350bc
et/sip/SipManager.java
|
| 0858806ffcb9ff34725abb79106aa1de27d1bf60 |
26-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Add Wifi High Perf. mode during a call.
To prevent the wifi from entering low-power mode due to the screen off
triggered by the proximity sensor.
Change-Id: I490bc594d800bc30c256e52ef3bce08bf86bc7b1
et/sip/SipAudioCallImpl.java
|
| 5424c8dcacf1c227fe7deb0185510614122ab447 |
25-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Add dynamic uid info for tracking the sip service usage.
Change-Id: Ibc340401b63799326b08aee6eba602a3e753b13f
et/sip/SipProfile.java
|
| 37f709aeb0424948a8f69577c6fad39dc95d7733 |
25-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "SipProfile: add isOutgoingCallAllowed() and new builder constructor" into gingerbread
|
| cf95f5d26363d4cd3815d31f5798f932a7720c17 |
23-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipProfile: add isOutgoingCallAllowed() and new builder constructor
Change-Id: I7ced47079fd2b00c7160b152eb4c1d34399e39dc
et/sip/SipProfile.java
|
| 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
et/sip/SipAudioCall.java
et/sip/SipAudioCallImpl.java
|
| b8790323473bef75a27d2da6fde2497b3bfe19eb |
19-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: fix few leaks when fail to add streams into a group.
Change-Id: Iefb3fe219ad48641da37a83c8d14e9ebf1d3086c
et/rtp/AudioGroup.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
et/sip/SipAudioCall.java
et/sip/SipAudioCallImpl.java
|
| ea4de5bd25b394a1bac6f27b43c4982aace2011e |
10-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipAudioCall: perform local ops before network op in endCall()
Change-Id: I1808f715d56c0979cea7741cb5bdb3831774d3ef
et/sip/SipAudioCallImpl.java
|
| 8e63ddb4c78dc4453d64ea6e94c109db703185e4 |
09-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: clean up unused class and fields.
Change-Id: I79ed7fb324fea9a52946340055b5ea1d389a926a
et/sip/BinderHelper.java
et/sip/SipManager.java
|
| cde66df44240cfe5a7bec12ac52464c3bf26c14f |
05-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Revert "Move SIP telephony related codes to framework."
This reverts commit b631dcf3eb449ddec756bea330f4e70b996ffb9e.
et/sip/SipManager.java
|
| 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
et/sip/SipManager.java
|
| 363c2ab82cca4f095e9e0c8465e28f6d27a24bf8 |
05-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Move the sip related codes to framework.
Change-Id: Ib81dadc39b73325c8438f078c7251857a83834fe
et/rtp/AudioCodec.java
et/rtp/AudioGroup.java
et/rtp/AudioStream.java
et/rtp/RtpStream.java
et/sip/BinderHelper.java
et/sip/ISipService.aidl
et/sip/ISipSession.aidl
et/sip/ISipSessionListener.aidl
et/sip/SdpSessionDescription.java
et/sip/SessionDescription.aidl
et/sip/SessionDescription.java
et/sip/SipAudioCall.java
et/sip/SipAudioCallImpl.java
et/sip/SipManager.java
et/sip/SipProfile.aidl
et/sip/SipProfile.java
et/sip/SipRegistrationListener.java
et/sip/SipSessionAdapter.java
et/sip/SipSessionState.java
|