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
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
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
SipURI returns port -1 when port is not present in the URI.
Don't call SipProfile.Builder.setPort() when that happens.
Change-Id: Ic5fe7301195705a77010038cae20d6629b33135e
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
* commit '9a8df8054b2e38a27d8e8e6b17365979218f0e3f':
docs: revise javadocs for sip add a package description, revise class descriptions and edit some method docs
(watch out auto-merge conflict for SipAudioCall).
Bug: 3113033, related CL: https://android-git/g/#change,75185
Change-Id: Ib48d3b990e229e0b341e47e10e76934e1a50d10f
Merge commit '3cb2d3be6cb501c77c7a5765d954363125857cca'
* commit '3cb2d3be6cb501c77c7a5765d954363125857cca':
SipService: supply PendingIntent when open a profile.
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
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
+ 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
+ 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
