17 Commits

Author SHA1 Message Date
Edward Lemur
c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00
Henrik Kjellander
a80c16a67c Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
This reverts commit c3771cc4d37f5573fe53b7c7cff295a4f0f9560f.
(breaks downstream internal project)

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2972463002 .
Cr-Commit-Position: refs/heads/master@{#18873}
2017-07-01 14:48:18 +00:00
kjellander
c3771cc4d3 Update includes for webrtc/{base => rtc_base} rename (2/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.

Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
2017-06-30 20:42:44 +00:00
deadbeef
1ee2125909 Adding PortAllocator option to support cases where sockets can't be bound.
This CL adds the flag "PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS", which will
force the creation of ports not bound to any specific network interface.
These are normally only used when network enumeration fails or is disabled,
but in some circumstances (such as the one the test case adds), they're the
only thing that works.

This will result in extra ports being gathered, which is why it's only enabled
behind a flag for now. In the future, we could probably introduce more
sophisticated "pruning" logic that would lessen the impact of the extra ports
when they're redundant, and make the flag the default.

Some other minor changes that were required to make this use case work:

* Allow a TCPPort to be used for outgoing connections even if it tries and
  fails to create a server socket.
* Allow Bind to fail if being called before Connect, and the IP is an "any"
  address (0.0.0.0 or ::), since this bind would have been mostly pointless
  anyway.
* Prevent P2PTransprotChannel from keeping a "backup" candidate pair using
  an "any address" network; we only want this for actual networks.

BUG=webrtc:7798

Review-Url: https://codereview.webrtc.org/2936553003
Cr-Commit-Position: refs/heads/master@{#18578}
2017-06-13 22:49:45 +00:00
deadbeef
f137e97adb Revert of Removing HTTPS and SOCKS proxy server code. (patchset #2 id:20001 of https://codereview.webrtc.org/2731673002/ )
Also needed to revert these CLs, which removed code used by the
code being un-removed:
https://codereview.webrtc.org/2745523004
https://codereview.webrtc.org/2754033003
https://codereview.webrtc.org/2758943002

Reason for revert:
This code is still being used by native application developers, so we should send a PSA announcing the deprecation and suggest an alternative before removing it.

Original issue's description:
> Removing HTTPS and SOCKS proxy server code.
>
> This isn't used any more so there's no point in maintaining it.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2731673002
> Cr-Commit-Position: refs/heads/master@{#17016}
> Committed: a1991c5175

TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2766063005
Cr-Commit-Position: refs/heads/master@{#17369}
2017-03-23 22:45:49 +00:00
deadbeef
a1991c5175 Removing HTTPS and SOCKS proxy server code.
This isn't used any more so there's no point in maintaining it.

BUG=None

Review-Url: https://codereview.webrtc.org/2731673002
Cr-Commit-Position: refs/heads/master@{#17016}
2017-03-03 18:23:08 +00:00
nisse
ede5da4960 Replace ASSERT by RTC_DCHECK in all non-test code.
Bulk of the changes were produced using

  git grep -l ' ASSERT(' | grep -v test | grep -v 'common\.h' |\
    xargs -n1 sed -i 's/ ASSERT(/ RTC_DCHECK(/'

followed by additional includes of base/checks.h in affected files,
and git cl format.

Also had to do some tweaks to #if !defined(NDEBUG) logic in the
taskrunner code (webrtc/base/task.cc, webrtc/base/taskparent.cc,
webrtc/base/taskparent.h, webrtc/base/taskrunner.cc), replaced to
consistently use RTC_DCHECK_IS_ON, and some of the checks needed
additional #if protection.

Test code was excluded, because it should probably use RTC_CHECK
rather than RTC_DCHECK.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2620303003
Cr-Commit-Position: refs/heads/master@{#16030}
2017-01-12 13:15:36 +00:00
hnsl
0483362377 Add disabled certificate check support to IceServer PeerConnection API.
Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS. Maintaining deprecated
backwards compatible support for "OPT_SSLTCP".

Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.

PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.

TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.

Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.

For security reasons we ensure that:

	1) The policy is always explicitly initialized to secure.
        2) API users have to explicitly integrate with the feature to
           use it, and will otherwise get no change in behavior.
	3) The feature is not immediately exposed in non-native
	   contexts. For example, disabling of certificate validation
           is not implemented via URI parsing since this would
           immediately allow it to be used from a web page.

This is a second attempt of https://codereview.webrtc.org/2557803002/
which was rolled back in https://codereview.webrtc.org/2590153002/

BUG=webrtc:6840

Review-Url: https://codereview.webrtc.org/2594623002
Cr-Commit-Position: refs/heads/master@{#15967}
2017-01-09 16:35:45 +00:00
magjed
d5236e2948 Revert of Add disabled certificate check support to IceServer PeerConnection API. (patchset #8 id:140001 of https://codereview.webrtc.org/2557803002/ )
Reason for revert:
This CL broke all Chromium WebRTC FYI bots. A roll+fix was attempted here: https://codereview.chromium.org/2590783003/, but failed to land. I'm reverting this CL now to make the tree green again. Make the API change gradual when you reland so that we can update Chromium between.

Original issue's description:
> Add disabled certificate check support to IceServer PeerConnection API.
>
> Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
> that it's not actually some kind of SSL over TCP. Also making it clear
> that it's mutually exclusive with OPT_TLS.
>
> Add "OPT_TLS_INSECURE" that implements the new certificate-check
> disabled TLS mode, which is also mutually exclusive with the other
> TLS options.
>
> PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
> the new insecure mode and added it as a RelayCredentials member.
>
> TurnPort: Add new TLS policy member with appropriate getter and setter
> to avoid constructor bloat. Initialize it from the RelayCredentials
> after the TurnPort is created.
>
> Expose the new feature in the PeerConnection API via
> IceServer.tls_certificate_policy as well as via the Android JNI
> PeerConnection API.
>
> For security reasons we ensure that:
>
> 	1) The policy is always explicitly initialized to secure.
>         2) API users have to explicitly integrate with the feature to
>            use it, and will otherwise get no change in behavior.
> 	3) The feature is not immediately exposed in non-native
> 	   contexts. For example, disabling of certificate validation
>            is not implemented via URI parsing since this would
>            immediately allow it to be used from a web page.
>
> BUG=webrtc:6840
>
> Review-Url: https://codereview.webrtc.org/2557803002
> Cr-Commit-Position: refs/heads/master@{#15670}
> Committed: b0f04fdb9e

TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,hnsl@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6840

Review-Url: https://codereview.webrtc.org/2590153002
Cr-Commit-Position: refs/heads/master@{#15703}
2016-12-20 10:22:06 +00:00
hnsl
b0f04fdb9e Add disabled certificate check support to IceServer PeerConnection API.
Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS.

Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.

PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.

TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.

Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.

For security reasons we ensure that:

	1) The policy is always explicitly initialized to secure.
        2) API users have to explicitly integrate with the feature to
           use it, and will otherwise get no change in behavior.
	3) The feature is not immediately exposed in non-native
	   contexts. For example, disabling of certificate validation
           is not implemented via URI parsing since this would
           immediately allow it to be used from a web page.

BUG=webrtc:6840

Review-Url: https://codereview.webrtc.org/2557803002
Cr-Commit-Position: refs/heads/master@{#15670}
2016-12-19 12:10:30 +00:00
johan
57e13defc7 Minor cleanup of rtc::BasicPacketSocketFactory implementation.
Remove unnecessary rtc:: namespace prefixes. Add #include <string>.

BUG=None

Review-Url: https://codereview.webrtc.org/2427413004
Cr-Commit-Position: refs/heads/master@{#14777}
2016-10-25 17:15:14 +00:00
kwiberg
3ec4679dd2 Replace scoped_ptr with unique_ptr in webrtc/p2p/
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1923163003

Cr-Commit-Position: refs/heads/master@{#12532}
2016-04-27 14:22:58 +00:00
Peter Boström
0c4e06b4c6 Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
2015-10-07 10:23:32 +00:00
pkasting@chromium.org
332331fb01 Use uint16s for port numbers in webrtc/p2p/base.
This is a necessary precursor to using uint16s for port numbers more
consistently in Chromium code.

This also makes some minor formatting changes in surrounding code (function declaration wrapping, virtual -> override).

BUG=chromium:81439
TEST=none
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7656 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 20:19:22 +00:00
henrike@webrtc.org
269fb4bc90 move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:20:11 +00:00
henrike@webrtc.org
28100cb388 Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
BUG=N/A
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 22:03:39 +00:00
henrike@webrtc.org
d1ba6d9cbf Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 17:30:28 +00:00