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gerrit-public.fairphone.software
/
platform
/
frameworks
/
base
/
7eb805707fc512f1711fbf00eb5eec2472a0bae1
/
voip
09225a4
am c0f2d952: am 0b7d6de1: Fix the build.
by repo sync
· 14 years ago
6b762aa
resolved conflicts for merge of 2b652855 to master
by Chia-chi Yeh
· 14 years ago
0b7d6de
Fix the build.
by repo sync
· 14 years ago
84f7f6b
SIP: Make SipAudioCallImpl use SimpleSessionDescription instead of javax.sdp.
by repo sync
· 14 years ago
e6c0c10
SDP: Add a simple class to help manipulate session descriptions.
by Chia-chi Yeh
· 14 years ago
7a69aef
RTP: Add log throttle for "no data".
by repo sync
· 14 years ago
4033a67
RTP: Update native part to reflect the API change.
by Chia-chi Yeh
· 14 years ago
37adc52
RTP: Add two getters to retrieve the current configuration from AudioStream.
by Chia-chi Yeh
· 14 years ago
32e106b
RTP: Extend codec capability and update the APIs.
by Chia-chi Yeh
· 14 years ago
5491c7a
am 852e5354: am 8544560c: SipPhone: fix missing-call DisconnectCause feedback
by Hung-ying Tyan
· 14 years ago
8544560
SipPhone: fix missing-call DisconnectCause feedback
by Hung-ying Tyan
· 14 years ago
2417132
am 37d1b249: am 97963794: SIP: convert enum to static final int.
by Hung-ying Tyan
· 14 years ago
9796379
SIP: convert enum to static final int.
by Hung-ying Tyan
· 14 years ago
03d8a8f
resolved conflicts for merge of cf1620d6 to master
by Hung-ying Tyan
· 14 years ago
c4b8747
SIP: add config flag for wifi-only configuration.
by Hung-ying Tyan
· 14 years ago
e30fc66
Fix build.
by Hung-ying Tyan
· 14 years ago
93f3947
Fix build
by Hung-ying Tyan
· 14 years ago
6c62609
resolved conflicts for merge of 394d1e4b to master
by Hung-ying Tyan
· 14 years ago
afa583e
SipAudioCall: expose startAudio()
by Hung-ying Tyan
· 14 years ago
6308514
am 156edcc9: am 9404e633: Merge "Add timer to SIP session creation process." into gingerbread
by Hung-ying Tyan
· 14 years ago
9352cf1
Add timer to SIP session creation process.
by Hung-ying Tyan
· 14 years ago
3cd497c
am 6962aff3: am 698ddb0c: Merge "Fix links in SIP API javadoc." into gingerbread
by Hung-ying Tyan
· 14 years ago
3e5246b
am 8a33e964: am 5306e0a8: Merge "SIP: add PEER_NOT_REACHABLE error feedback." into gingerbread
by Hung-ying Tyan
· 14 years ago
286bb5a
Fix links in SIP API javadoc.
by Hung-ying Tyan
· 14 years ago
fbfb6f2
am 188a8f1b: am 89a7180a: Merge "SipService: ignore connect event for non-active networks." into gingerbread
by Hung-ying Tyan
· 14 years ago
b1c4a01
am ae83faa3: am 13f6270e: SipAudioCall: use SipErrorCode instead of string in onError()
by Hung-ying Tyan
· 14 years ago
fcc7fb8
am 3692af92: am 99bf4e45: SIP: remove dependency on javax.sip
by Hung-ying Tyan
· 14 years ago
074663c
am ca83c25d: am 4565933f: Merge "SipService: deliver connectivity change to all sessions." into gingerbread
by Hung-ying Tyan
· 14 years ago
ae076d3
SIP: add PEER_NOT_REACHABLE error feedback.
by Hung-ying Tyan
· 14 years ago
12bec5d
SipService: ignore connect event for non-active networks.
by Hung-ying Tyan
· 14 years ago
13f6270
SipAudioCall: use SipErrorCode instead of string in onError()
by Hung-ying Tyan
· 14 years ago
99bf4e4
SIP: remove dependency on javax.sip
by Hung-ying Tyan
· 14 years ago
d231aa8
SipService: deliver connectivity change to all sessions.
by Hung-ying Tyan
· 14 years ago
0b5a8bd
am a5dce0c1: am 3d7606aa: SIP: enhance timeout and registration status feedback.
by Hung-ying Tyan
· 14 years ago
3d7606a
SIP: enhance timeout and registration status feedback.
by Hung-ying Tyan
· 14 years ago
e765994
am 38dc67f4: am 25b52a2f: SIP: remove dependency on javax.sip.SipException.
by Hung-ying Tyan
· 14 years ago
25b52a2
SIP: remove dependency on javax.sip.SipException.
by Hung-ying Tyan
· 14 years ago
a97ccc0
am 5f93c39c: am ca3c24db: Merge "SIP: add SipErrorCode for error feedback." into gingerbread
by Hung-ying Tyan
· 14 years ago
903e103
SIP: add SipErrorCode for error feedback.
by Hung-ying Tyan
· 14 years ago
3384e0f
am c2eff4a7: am f6936a3a: Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread
by Chia-chi Yeh
· 14 years ago
f6936a3a
Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread
by Chia-chi Yeh
· 14 years ago
557b04d
RTP: prevent buffer overflow in AudioRecord.
by Chia-chi Yeh
· 14 years ago
3559d1d
am ea16e72b: am b355714a: Merge "SipManager: always return true for SIP API and VOIP support query." into gingerbread
by Hung-ying Tyan
· 14 years ago
643fce9
SipManager: always return true for SIP API and VOIP support query.
by Hung-ying Tyan
· 14 years ago
3d67c56
resolved conflicts for merge of 12eaf9d5 to master
by repo sync
· 14 years ago
dc296b0
Merge "SipService: reduce the usage of javax.sdp.*." into gingerbread
by Chia-chi Yeh
· 14 years ago
95b15c3
SipService: reduce the usage of javax.sdp.*.
by Chia-chi Yeh
· 14 years ago
e93a492
am 18dfae22: am 60264b30: SipProfile: remove outgoingCallAllowed flag.
by Hung-ying Tyan
· 14 years ago
60264b30
SipProfile: remove outgoingCallAllowed flag.
by Hung-ying Tyan
· 14 years ago
f83d4f1
resolved conflicts for merge of 3e4975a5 to master
by Hung-ying Tyan
· 14 years ago
3424c02
Add software features for SIP and VOIP
by Hung-ying Tyan
· 14 years ago
b6d4723
Revert "Add Wifi High Perf. mode during a call."
by Chung-yih Wang
· 14 years ago
0858806
Add Wifi High Perf. mode during a call.
by Chung-yih Wang
· 14 years ago
14e0062
Merge "Revert "RTP: integrate the echo canceller from speex."" into gingerbread
by Chia-chi Yeh
· 14 years ago
7fa7ee1
Revert "RTP: integrate the echo canceller from speex."
by Chia-chi Yeh
· 14 years ago
5424c8d
Add dynamic uid info for tracking the sip service usage.
by Chung-yih Wang
· 14 years ago
37f709a
Merge "SipProfile: add isOutgoingCallAllowed() and new builder constructor" into gingerbread
by Hung-ying Tyan
· 14 years ago
cf95f5d
SipProfile: add isOutgoingCallAllowed() and new builder constructor
by Hung-ying Tyan
· 14 years ago
3294d44
Add confcall management to SIP calls
by Hung-ying Tyan
· 14 years ago
4ae6ec4
RTP: integrate the echo canceller from speex.
by Chia-chi Yeh
· 14 years ago
2880ef8
RTP: reduce the latency by overlapping AudioRecord and AudioTrack.
by Chia-chi Yeh
· 14 years ago
b879032
RTP: fix few leaks when fail to add streams into a group.
by Chia-chi Yeh
· 14 years ago
3459d30
RTP: remove froyo-compatible code.
by Chia-chi Yeh
· 14 years ago
cfd15dd
Fix the IN_CALL mode issue.
by Chung-yih Wang
· 14 years ago
ea4de5bd
SipAudioCall: perform local ops before network op in endCall()
by Hung-ying Tyan
· 14 years ago
8e63ddb
SIP: clean up unused class and fields.
by Hung-ying Tyan
· 14 years ago
4c5d28c
RTP: move into frameworks.
by Chia-chi Yeh
· 14 years ago
cde66df
Revert "Move SIP telephony related codes to framework."
by Chung-yih Wang
· 14 years ago
b631dcf
Move SIP telephony related codes to framework.
by Chung-yih Wang
· 14 years ago
363c2ab
Move the sip related codes to framework.
by Chung-yih Wang
· 14 years ago