This CL moves candidate.h into the public API, since it has
been implicitly included before.
This is a straightforward way of solving the circular
dependencies involving that file. For instance,
libjingle_peerconnection_api includes candidate.h from
jsepicecandidate.h, but _api can't depend on rtc_p2p, which
depends on _api. In fact, _api can't depend on much at all
since it's a very high level abstraction; instead, things
should depend on it.
Furthermore, we have the case where deprecated headers
include headers in internal modules. I just have to turn
off include checking for those, but that's not a big deal.
This CL punts the problem of callfactoryinterface.h being
implicitly included, and pulling in most of the call
module with it. This should be addressed in a follow-up
CL.
Bug: webrtc:7504
Change-Id: I1b1729408158418333ccdf702bf529386090f0d7
Reviewed-on: https://webrtc-review.googlesource.com/2020
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20034}
If the network interface appears active, but binding the sockets fails,
then it won't produce any candidates even though it's never marked as
"network failed". So this was causing nothing to happen once a network
change event occurs and the interface becomes usable again.
So, this CL adds the condition that we only disable gathering of local
ports if we don't have them already.
See bug for more details.
BUG=webrtc:8256
Review-Url: https://codereview.webrtc.org/3015543002
Cr-Commit-Position: refs/heads/master@{#20007}
The SrtpTransport takes the SRTP responsibilities from the BaseChannel
and SrtpFilter. SrtpTransport is now responsible for setting the crypto
keys, protecting and unprotecting the packets. SrtpTransport doesn't
know if the keys are from SDES or DTLS handshake.
BaseChannel is now only responsible setting the offer/answer for SDES
or extracting the key from DtlsTransport and configuring the
SrtpTransport.
SrtpFilter is used by BaseChannel as a helper for SDES negotiation.
BUG=webrtc:7013
Change-Id: If61489dfbdf23481a1f1831ad181fbf45eaadb3e
Reviewed-on: https://webrtc-review.googlesource.com/2560
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19977}
The support of fallback from DTLS to SDES is removed in this CL.
Setting an SDP with both DTLS fingerprint and SDES crypto would fail.
BUG=webrtc:8266
Review-Url: https://codereview.webrtc.org/3011133002
Cr-Commit-Position: refs/heads/master@{#19903}
Reason for revert:
This seems to be causing some video freezes. See https://bugs.chromium.org/p/webrtc/issues/detail?id=8251
Original issue's description:
> Completed the functionalities of SrtpTransport.
>
> The SrtpTransport takes the SRTP responsibilities from the BaseChannel
> and SrtpFilter. SrtpTransport is now responsible for setting the crypto
> keys, protecting and unprotecting the packets. SrtpTransport doesn't know
> if the keys are from SDES or DTLS handshake.
>
> BaseChannel is now only responsible setting the offer/answer for SDES
> or extracting the key from DtlsTransport and configuring the
> SrtpTransport.
>
> SrtpFilter is used by BaseChannel as a helper for SDES negotiation.
>
> BUG=webrtc:7013
>
> Review-Url: https://codereview.webrtc.org/2997983002
> Cr-Commit-Position: refs/heads/master@{#19636}
> Committed: e683c6871fTBR=deadbeef@webrtc.org,pthatcher@google.com,zhihuang@webrtc.org
Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/3018513002
Cr-Commit-Position: refs/heads/master@{#19895}
I'm not sure why we ever had this in the first place, and it confuses
people on a nearly weekly basis, so let's get rid of it. The protocols
are enabled right after the corresponding gathering is done, so the only
real effect it has is to produce confusing log messages (first
"candidate not signaled because protocol not enabled", then "protocol
enabled, signaling candidate" right afterwards).
BUG=None
Review-Url: https://codereview.webrtc.org/3018483002
Cr-Commit-Position: refs/heads/master@{#19873}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
