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
The previous implementation registers receivers when SipService starts up.
If the user doesn't use SIP at all, SipService will still process connecivity
and wifi state change events, which involves holding wake lock and thus
consumes power unnecessarily.
With this CL, SipService is completely idle if the user doesn't use SIP at all.
It registers receivers only when at least one account is opened.
Bug: 3326998
Change-Id: Idea43747f8204b0ccad3fc05a1b1c0b29c9b2557
Originally a stream does not send packets when it is receive-only or there is
nothing to mix. However, this causes some problems with certain firewalls and
proxies. A firewall might remove a port mapping when there is no outgoing
packet for a preiod of time, and a proxy might wait for incoming packets from
both sides before start forwarding. To solve these problems, we send out a
silence packet on the stream for every second. It should be good enough to
keep the stream alive with relatively low resources.
Bug: 3119690
Change-Id: Ib9c55e5dddfba28928bd9b376832b68bda24c0e4
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
+ Also fix race between ending and changing (holding/unholding) a SIP call.
+ Remove an unused method.
Bug : 3128233
Change-Id: Ie18d8333a88f0d9906d54988243d909b58e07e4b
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
Make it return true for all existing accounts.
Rename mOpened to mOpenedToReceiveCalls to make it less confusing.
Bug: 3155849
Change-Id: I327f411bf76afd73434ad1fa2ffef3db1e35d778
* commit '9a8df8054b2e38a27d8e8e6b17365979218f0e3f':
docs: revise javadocs for sip add a package description, revise class descriptions and edit some method docs
Merge commit 'd4d3f36f4c25b41f4253eadd5e67035fe220cad3'
* commit 'd4d3f36f4c25b41f4253eadd5e67035fe220cad3':
Clean up pending sessions on incoming call in SipService
Rewrite using integer arithmetic to get full 32-bit precision instead
of 23-bit in single precision floating-points.
Bug: 3029745
Change-Id: If67dcc403923755f403d08bbafb41ebce26e4e8b
