Commit Graph

11 Commits

Author SHA1 Message Date
Hung-ying Tyan
97963794af SIP: convert enum to static final int.
Converts SipErrorCode and SipSessionState.

Change-Id: Iee3a465649ea89d395b2336bbd673c25113e5f93
2010-09-20 09:51:31 +08:00
Hung-ying Tyan
afa583e655 SipAudioCall: expose startAudio()
so that apps can start audio when time is right.

Change-Id: I7ae96689d3a8006b34097533bc2434bc3814b82a
2010-09-17 15:58:18 +08:00
Hung-ying Tyan
9352cf1a4d 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
2010-09-17 13:31:05 +08:00
Hung-ying Tyan
286bb5a00b Fix links in SIP API javadoc.
Change-Id: I839280fe18502bb576f6e9c9a7948077c02fa570
2010-09-16 03:52:10 +08:00
Hung-ying Tyan
13f6270eb1 SipAudioCall: use SipErrorCode instead of string in onError()
and fix callback in setListener().

Change-Id: Ic2622df992a2ad45cb1e3f71736f320897ae8fb3
2010-09-14 21:36:10 +08:00
Hung-ying Tyan
25b52a2f97 SIP: remove dependency on javax.sip.SipException.
Change-Id: I77d289bef1b5e7f1ec0c0408d0bbf96c21085cd7
2010-09-13 16:50:12 +08:00
Hung-ying Tyan
903e103160 SIP: add SipErrorCode for error feedback.
Change-Id: I8b071d4933479b780a403d0bfa30511f4c23ca8f
2010-09-10 17:15:06 +08:00
Chia-chi Yeh
95b15c3560 SipService: reduce the usage of javax.sdp.*.
After this change, SipAudioCallImpl is the only place still using it.

Change-Id: I5693bffa54f9e19cbfa70b45dfcf40fba04dedbb
2010-09-02 22:15:26 +08:00
Hung-ying Tyan
3294d44b96 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
2010-08-24 17:54:47 +08:00
Chung-yih Wang
cfd15dd3c8 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
2010-08-16 18:02:31 +08:00
Chung-yih Wang
363c2ab82c Move the sip related codes to framework.
Change-Id: Ib81dadc39b73325c8438f078c7251857a83834fe
2010-08-05 10:25:53 +08:00