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gerrit-public.fairphone.software
/
platform
/
frameworks
/
base
/
0ee2431b27b0a0a81ad2f967723dee627a8517a9
/
voip
ee59e6a
SipService: handle connectivity changes correctly.
by Chia-chi Yeh
· 13 years ago
6d8b9b8
Merge "RTP: Update parameters for larger packet intervals."
by Chia-chi Yeh
· 13 years ago
7a685e8
Merge "SIP: fix keep-alive measurement and increase the timeout."
by Chia-chi Yeh
· 13 years ago
d17b6d5
SIP: fix keep-alive measurement and increase the timeout.
by Chia-chi Yeh
· 13 years ago
be57bfe
RTP: Update parameters for larger packet intervals.
by Chia-chi Yeh
· 13 years ago
81a5ec5
Merge "RTP: support payloads with larger packetization interval."
by Chia-chi Yeh
· 13 years ago
fa6067f
Merge "VoIP JNI: Force AEC on for tuna board"
by Eric Laurent
· 13 years ago
35d05dc
RTP: support payloads with larger packetization interval.
by Chia-chi Yeh
· 13 years ago
54eabd6
SIP: avoid extreme small values in Min-Expires headers.
by Chia-chi Yeh
· 13 years ago
74e0a99
VoIP JNI: Force AEC on for tuna board
by Eric Laurent
· 13 years ago
5f76006
SIP: add the check for expiry time in Contact header.
by Chia-chi Yeh
· 13 years ago
dc5bbe9
Handle SIP authentication response for BYE.
by Hung-ying Tyan
· 13 years ago
53ad2c7
am 0793586b: am f8c1f129: am e1d27154: am f87743e7: Merge "Prevent NullPointerException cases while using SipService."
by Conley Owens
· 13 years ago
0793586
am f8c1f129: am e1d27154: am f87743e7: Merge "Prevent NullPointerException cases while using SipService."
by Conley Owens
· 13 years ago
25ccbb9
Prevent NullPointerException cases while using SipService.
by Masahiko Endo
· 13 years ago
5fb3ba6
Issue 3370834: No Echo canceler for SIP
by Eric Laurent
· 13 years ago
307f15f
Add REFER handling.
by repo sync
· 13 years ago
3eeb1a9
Merge "Keep last known keepalive interval to avoid duplicate effort."
by Hung-ying Tyan
· 13 years ago
9324e04
Merge "Do not hold wifi lock when SIP is also available over mobile network."
by Hung-ying Tyan
· 13 years ago
f8c34ad
Merge "Do not keep alive for re-established call."
by Hung-ying Tyan
· 13 years ago
9edfa10
Keep last known keepalive interval to avoid duplicate effort.
by Hung-ying Tyan
· 13 years ago
8ba4566
Do not keep alive for re-established call.
by Hung-ying Tyan
· 13 years ago
f89654d
Do not hold wifi lock when SIP is also available over mobile network.
by Hung-ying Tyan
· 13 years ago
a6cec8f
Synchronize SipWakeupTimer.onReceive()
by Hung-ying Tyan
· 13 years ago
129d0b0
Make NAT port timeout measurement more flexible.
by Hung-ying Tyan
· 13 years ago
99705b5
Record external IP and port from SIP responses
by Hung-ying Tyan
· 13 years ago
2093561
Support INVITE w/o SDP.
by repo sync
· 13 years ago
233718c
Start keepalive process for the caller of a SIP call
by Hung-ying Tyan
· 13 years ago
1aceda3
Support Invite w/ Replaces request.
by repo sync
· 13 years ago
e65f3a8
Restart NAT port timeout measurement when keepalive fails and other fixes
by Hung-ying Tyan
· 13 years ago
4af085f
Execute all the due wakeup events in SipWakeupTimer.
by Hung-ying Tyan
· 13 years ago
1275070
Keep the keepalive process going after NAT port is changed.
by Hung-ying Tyan
· 13 years ago
4a267a9
Move the keepalive process to SipSessionImpl and make it reusable.
by Hung-ying Tyan
· 14 years ago
ac320b2
Merge "Move WakeupTimer out of SipService."
by Hung-ying Tyan
· 14 years ago
5621554
Move WakeupTimer out of SipService.
by Hung-ying Tyan
· 14 years ago
c133781
Fix the issue of onNetwork in UI thread.
by repo sync
· 14 years ago
bb0a989
Add KeepAlive Interval Measurement.
by Chung-yih Wang
· 14 years ago
34bb419
update for new audio.h header location
by Dima Zavin
· 14 years ago
4b3913a
Squashed commit of the following:
by Andreas Huber
· 14 years ago
b8df57d
am d81214da: am a7a9c4cb: am 46524f83: Merge "docs: add package description for RTP" into honeycomb-mr1
by Scott Main
· 14 years ago
d81214d
am a7a9c4cb: am 46524f83: Merge "docs: add package description for RTP" into honeycomb-mr1
by Scott Main
· 14 years ago
de9acb7
docs: add package description for RTP
by Scott Main
· 14 years ago
24fc2fb
audio/media: convert to using the audio HAL and new audio defs
by Dima Zavin
· 14 years ago
d8cbd16
am 7a492a9a: am b7a76e84: am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread
by Eric Laurent
· 14 years ago
7a492a9
am b7a76e84: am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread
by Eric Laurent
· 14 years ago
b7a76e8
am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread
by Eric Laurent
· 14 years ago
d7a724e
Issue 4157048: mic gain for VoIP/SIP calls.
by Eric Laurent
· 14 years ago
397de16
am fae5e289: am 6f67e7bf: am 2e383bc6: Merge "Making it possible to call SIP calls with special allowed chars."
by Brad Fitzpatrick
· 14 years ago
fae5e28
am 6f67e7bf: am 2e383bc6: Merge "Making it possible to call SIP calls with special allowed chars."
by Brad Fitzpatrick
· 14 years ago
b5c72ea
Making it possible to call SIP calls with special allowed chars.
by Magnus Strandberg
· 14 years ago
3f9e089
Include strings.h instead of string.h for the strcasecmp prototype.
by Carl Shapiro
· 14 years ago
3070af0
frameworks/base: remove LOCAL_PRELINK_MODULE
by Iliyan Malchev
· 14 years ago
6defd2d
NEW_API: Unhide RTP APIs.
by Chia-chi Yeh
· 14 years ago
c52f5b2
RTP: update javadocs.
by Chia-chi Yeh
· 14 years ago
89bc1fe
Activate the wifi high perf. for sip calls.
by Chung-yih Wang
· 14 years ago
fcd0e50
Add rport argument for a reinvite request.
by Chung-yih Wang
· 14 years ago
9e25df4
Make SIP AuthName APIs public.
by Chung-yih Wang
· 14 years ago
2ba92c7
do not merge bug 3370834 Cherrypick from master
by Jean-Michel Trivi
· 14 years ago
14b6d06
Merge changes Ib70e0cf2,I0691cd70 into gingerbread
by Hung-ying Tyan
· 14 years ago
f46013b
Merge "Merge "SipService: registers broadcast receivers on demand."" into honeycomb
by Hung-ying Tyan
· 14 years ago
e9b5407
Merge "SipService: registers broadcast receivers on demand."
by Hung-ying Tyan
· 14 years ago
40f2cac
Merge "SipService: release wake lock for cancelled tasks."
by Hung-ying Tyan
· 14 years ago
0f7de88
Merge "Add auth. username in SipProfile." from gingerbread.
by Chung-yih Wang
· 14 years ago
f268a2f
Add auth. username in SipProfile.
by Chung-yih Wang
· 14 years ago
f0bb1ce
SipService: registers broadcast receivers on demand.
by Hung-ying Tyan
· 14 years ago
d87be27
Enable built-in echo canceler if available.
by Chia-chi Yeh
· 14 years ago
4bf82df
Do not set back to AudioManager.MODE_NORMAL in SipAudioCall.
by Chia-chi Yeh
· 14 years ago
0c01e6e
SipService: release wake lock for cancelled tasks.
by Hung-ying Tyan
· 14 years ago
d0da380
am dc78e3fe: am 3cf71376: RTP: Send silence packets on idle streams for every second.
by Chia-chi Yeh
· 14 years ago
3cf7137
RTP: Send silence packets on idle streams for every second.
by Chia-chi Yeh
· 14 years ago
33808c6
am aec9a33f: am e0bd2688: Merge "Check if VoIP API is supported in SipManager." into gingerbread
by Hung-ying Tyan
· 14 years ago
5bd3782
Check if VoIP API is supported in SipManager.
by Hung-ying Tyan
· 14 years ago
635b2b7
am d90bc225: am a936b256: Remove SIP realm/domain check
by Hung-ying Tyan
· 14 years ago
a936b25
Remove SIP realm/domain check
by Hung-ying Tyan
· 14 years ago
58ee2ac
Check port in create peer's SIP profile.
by Hung-ying Tyan
· 14 years ago
eecf4a6
Check port in create peer's SIP profile.
by Hung-ying Tyan
· 14 years ago
c030a16
am c9cc9ab5: am 5f86d7f5: Merge "Fix SIP bug of different transport/port used for requests." into gingerbread
by Chung-yih Wang
· 14 years ago
f053292
Fix SIP bug of different transport/port used for requests.
by Chung-yih Wang
· 14 years ago
2aef9a1
am 7da1ffc9: am e2abd103: Merge "Set AudioGroup mode according to audio settings" into gingerbread
by Hung-ying Tyan
· 14 years ago
e2abd10
Merge "Set AudioGroup mode according to audio settings" into gingerbread
by Hung-ying Tyan
· 14 years ago
d6b0d68
am 6034f9b2: am 06e8cdc0: Fix race between ending and answering a SIP call.
by Hung-ying Tyan
· 14 years ago
db42452
am ed34b244: am d7116ff1: Merge "Do not suppress error feedback during a SIP call." into gingerbread
by Hung-ying Tyan
· 14 years ago
06e8cdc
Fix race between ending and answering a SIP call.
by Hung-ying Tyan
· 14 years ago
4c7cc83
Merge "RTP: Prepare to unhide the APIs."
by Chia-chi Yeh
· 14 years ago
53aa6ef
RTP: Prepare to unhide the APIs.
by Chia-chi Yeh
· 14 years ago
1c8c173
am c41b27e2: am 349f3509: Merge "Correct SipService.isOpened() implementation." into gingerbread
by Hung-ying Tyan
· 14 years ago
1210067
am 5c85338d: am d9e12303: Merge "Notify SipSessions before closing SIP stack." into gingerbread
by Hung-ying Tyan
· 14 years ago
ebf28fa
am 0e58a952: am 0bba9535: Merge "Throw proper exceptions in SipManager" into gingerbread
by Hung-ying Tyan
· 14 years ago
342a9be
am e843dfa8: am bd399b0b: Merge "RTP: Pause echo suppressor when far-end volume is low." into gingerbread
by Chia-chi Yeh
· 14 years ago
fa81463
Set AudioGroup mode according to audio settings
by Hung-ying Tyan
· 14 years ago
4189d99
Do not suppress error feedback during a SIP call.
by Hung-ying Tyan
· 14 years ago
349f3509
Merge "Correct SipService.isOpened() implementation." into gingerbread
by Hung-ying Tyan
· 14 years ago
d9e1230
Merge "Notify SipSessions before closing SIP stack." into gingerbread
by Hung-ying Tyan
· 14 years ago
0bba953
Merge "Throw proper exceptions in SipManager" into gingerbread
by Hung-ying Tyan
· 14 years ago
bd399b0
Merge "RTP: Pause echo suppressor when far-end volume is low." into gingerbread
by Chia-chi Yeh
· 14 years ago
8d1b2a1
Throw proper exceptions in SipManager
by Hung-ying Tyan
· 14 years ago
262cdfc
Correct SipService.isOpened() implementation.
by Hung-ying Tyan
· 14 years ago
e5bc8f6
am 9a8df805: am 1112632a: Merge "docs: revise javadocs for sip add a package description, revise class descriptions and edit some method docs" into gingerbread
by Scott Main
· 14 years ago
02b1d68
docs: revise javadocs for sip
by Scott Main
· 14 years ago
0c7d306
RTP: Pause echo suppressor when far-end volume is low.
by Chia-chi Yeh
· 14 years ago
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