and fix how SipErrorCode.SERVER_ERROR is determinted from server response, not
from local exceptions.
http://b/issue?id=3041332
Change-Id: Idce67e29858d5c7573b98b7fa1fac074913d71d6
Merge commit '245475925eff61ee76bde58de69253a889e39d0a' into gingerbread-plus-aosp
* commit '245475925eff61ee76bde58de69253a889e39d0a':
Fix the startAudio order for 3-way calls.
Merge commit '3234652242f54e3366e7c74e5a0cf0a7da5871b4' into gingerbread-plus-aosp
* commit '3234652242f54e3366e7c74e5a0cf0a7da5871b4':
Don't enter DISCONNECTING state when the call/connection is not alive
+ check REQUEST_TERMINATED response on INVITE not CANCEL,
+ check if a TransactionTerminatedEvent matches the ongoing transaction,
+ add log to track SipConnection disconnect events.
Change-Id: I28325be62ac44e4a7507d3c4b5b78b066c0ea2ad
and add new CROSS_DOMAIN_AUTHENTICATION error code and OUT_OF_NETWORK
DisconnectCause.
http://b/issue?id=3020185
Change-Id: Icc0a341599d5a72b7cb2d43675fbddc516544978
Merge commit 'bd2294204e3edaede3fe81eb9b11c05c4fafe627' into gingerbread-plus-aosp
* commit 'bd2294204e3edaede3fe81eb9b11c05c4fafe627':
Fix the unhold issue especially if one is behind NAT.
As per SMS specification in 3GPP2 C.S0015-B, section 4.3.1.6, if the
Subaddress is included in a CDMA SMS message, it needs to be used for
duplicate detection. Subaddress, which is an optional field was omitted
while computing the SMS fingerprint. Hence it was never being used in
duplicate detection if it was included in the SMS. Add subaddress to the
SMS fingerprint.
Change-Id: Iad9e89887a17caba59033ab8f8d94b63b33cb4bb
Merge commit '194bbcce9ba15634500f542b9ea017b2cf154b45' into gingerbread-plus-aosp
* commit '194bbcce9ba15634500f542b9ea017b2cf154b45':
SIP: longer timeout for making call, shorter for cancelling
Merge commit '84a357bb6a8005e1c5e924e96a8ecf310e77c47c' into gingerbread-plus-aosp
* commit '84a357bb6a8005e1c5e924e96a8ecf310e77c47c':
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
+fix the unknown call flash for answering an incoming call and
updating the screen if the background call got dropped.
+change the getFirstActiveBgCall to return the call if the state
is not IDLE. This will help to fix unknown flash if the background
call got dropped.
Change-Id: I9803ccebd919acbd5296e7dfde7dc5f29cc9f180
Merge commit 'ee2ef3220fd27a6332acb2f65951a7fe91e9dfa6' into gingerbread-plus-aosp
* commit 'ee2ef3220fd27a6332acb2f65951a7fe91e9dfa6':
Use PhoneBase in the phone list.
In 1x, if the data call is torn down before radio power off, modem will
have to send a data call release and change to initialization state followed
by idle state and send out power down registration. If the power off request is sent
to the modem during Initialization state after call release, there is a chance that
modem does not perform power down registration.
Instead if we directly initiate a power down, modem just sets a power down registration
bit in the release order. This change also optimizes the power down procedure in 1x by
letting the modem handle data call release during power down.
Change-Id: I0f083cc3b005ec1e64105350abb43d10583b0881
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
Merge commit '97963794af1e18674dd111e3ad344d90b16c922c' into gingerbread-plus-aosp
* commit '97963794af1e18674dd111e3ad344d90b16c922c':
SIP: convert enum to static final int.
For bug 3001613.
Only use PhoneBase (not PhoneProxy) in CallManager.
Both PhoneBase and PhoneProxy implement Phone interface,
such as dial(). The real implementation, for
example in GSM, is in GSMPhone extending from PhoneBase.
So that foregroundCall.getPhone() returns GSMPhone obj. On the other hand,
PhoneFactory.getDefaultPhone() returns PhoneProxy obj, which has a class
member of GSMPhone.
Therefore for phone returned by PhoneFacotry, which is used by PhoneApp,
phone.getForegroundCall().getPhone() != phone
Change-Id: I8a304098dd86762aaee56fb3c8b76c883e8c9a4f
Merge commit '1d1583573d2099756bbbeef48d97c280edc393e0' into gingerbread-plus-aosp
* commit '1d1583573d2099756bbbeef48d97c280edc393e0':
SipPhone: do not append SIP domain to PSTN number
+ 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
Merge commit 'd8f3d167353f6c6f6c5cb7a4c8e941c03b8e9511' into gingerbread-plus-aosp
* commit 'd8f3d167353f6c6f6c5cb7a4c8e941c03b8e9511':
Add a new phone state ANSWERING.
The state ANSWERING is set when we answer an incoming sip call, i.e.
sending a 'OK' response to the peer. The state will be set to ACTIVE
once the 'ACK' from peer is received.
Change-Id: I84ee3cc68222eb34e032896ce23f7431d4ad774a
Merge commit '968735e580b7728c6a17aff9b7356f9d74dc6eb3' into gingerbread-plus-aosp
* commit '968735e580b7728c6a17aff9b7356f9d74dc6eb3':
Fixing the NPE in SipPhone
