This patch adds TurnPort::Release that release a TURN allocation
by sending a REFRESH with lifetime 0 without destroying the object.
This allows for graceful shutdown of a TurnPort that can e.g be used
for mobility.
Bug: webtrc:9067
Change-Id: I1e4d9232ae08d6fe14f5612f776a541c03c3beec
Reviewed-on: https://webrtc-review.googlesource.com/64722
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22666}
IceConfig contains a set of parameters that affect the behavior of ICE.
Inconsistent or conflicting parameters lead to erroneous or
unpredicatble behavior in the network stack. Sanity checks are now added
to validate IceConfig.
TBR=magjed@webrtc.org
Bug: webrtc:8993
Change-Id: I708bc3f1ef970872754a82a47a509bda15061ca6
Reviewed-on: https://webrtc-review.googlesource.com/60847
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22664}
This patch modifies StunMessage to allow adding of attributes
in the 0x4000-0x7FFF range without adding them to stun.cc.
Before this patch this was allowed in the 0xC000-0xFFFF range
but the RFC specifies that both of these ranges are implementation
defined.
BUG=webrtc:8313
Change-Id: Ib74f5d02a06807aeca4fc3f1f3028271e233f004
Reviewed-on: https://webrtc-review.googlesource.com/64404
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22619}
conflict.
A role conflict received from an unknown address (peer reflexive
candidate) results in an early return before signaling the unknown
address to P2PTransportChannel. Without this signal, there is no
candidate pair or TURN entry created, and sending the error response
when handling the role conflict fails.
Bug: webrtc:9034
Change-Id: I0f1b232a574449e98025618d93aac8a91b30e14b
Reviewed-on: https://webrtc-review.googlesource.com/63840
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22588}
The convention is reinforced so that setting a rtc::Optional IceConfig
parameter to null restores the default value. Helper getters are added
to IceConfig to provide either user-defined value or the default.
Shared constants and config defaults used in p2p are moved to
p2pconstants.h/cc for future management with sanity checks.
Bug: webrtc:8993
Change-Id: I976cf1eef5a654b8911f449248bb2f3086279db8
Reviewed-on: https://webrtc-review.googlesource.com/61149
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22575}
Some implementations of std::max_element (used to find the "most
pingable" connection) seem to compare an element with itself, which
MorePingable doesn't handle.
Fixing by handling the self-comparison outside MorePingable.
Bug: webrtc:8697
Change-Id: Ieb34580f52037639c00041a4e65901cad92d0971
Reviewed-on: https://webrtc-review.googlesource.com/62402
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22543}
Also removing the implicit InterfaceAddress constructor that takes an
IPAddress, so that issues like this won't happen in the future.
And adding a convenience "Network::AddIP" method that takes an
IPAddress, so that code doing that (previously relying on the implicit
constructor) will continue to work.
Bug: webrtc:8972
Change-Id: Id5cf0fca481cfee3f8ab83412fcb41886535bba2
Reviewed-on: https://webrtc-review.googlesource.com/59461
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22504}
Add configurable parameters in RTCConfiguration with the default value
given by the constants CONNECTION_WRITE_CONNECT_TIME and
CONNECTION_WRITE_CONNECT_FAILURES in the ICE implementation. These two
parameters define the time period for which a candidate pair must wait
for ping response and the minimum number of connectivity checks that
the pair must send without response before its state becomes unreliable
from writable as defined in the current ICE implementation.
Bug: webrtc:8988
Change-Id: I484599b7d776489a87741ffea8926df766095da9
Reviewed-on: https://webrtc-review.googlesource.com/60704
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22411}
The connectivity check intervals for candidate pairs with strong and
weak connectivity are currently constants in the ICE implementation. A
set of suboptimal value of these constants for a given application may
result in undesirable behavior including excessive network switching
latency. This CL adds these intervals to RTCConfiguration that is
available to applications to configure, while maintaining the original
constants as their default value for compatibility with existing
applications.
Bug: webrtc:8988
Change-Id: I804b0f4cf7881be7d3c8aec2776bc9596de72482
Reviewed-on: https://webrtc-review.googlesource.com/60585
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22351}
candidate keepalive intervals.
StunStats for a STUN candidate cannot be updated after the initial report
in the stats collector. This is caused by the early return of cached
candidate reports for future queries after the initial report creation.
The STUN keepalive interval cannot be configured for UDPPort because of
incorrect type screening, where only StunPort was supported.
TBR=pthatcher@webrtc.org
Bug: webrtc:8951
Change-Id: I0c9c414f43e6327985be6e541e17b5d6f248a79d
Reviewed-on: https://webrtc-review.googlesource.com/58560
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22278}
is configured.
An immediate (re)sorting of candidate paris reduces the latency of
network switching when it is necessary in ICE after (re)configuring the
network preference. A fix of comment and boilerplate code is also
included.
Bug: None
Change-Id: I8685235172d97193ffa6b53d4d2c7796fd01f861
Reviewed-on: https://webrtc-review.googlesource.com/57340
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22197}
There was an implementation, but it relied on SSLCertificate::GetChain,
which was never implemented. Except in the fake certificate classes
used by the stats collector tests, hence the tests were passing.
Instead of implementing GetChain, we decided (in
https://webrtc-review.googlesource.com/c/src/+/6500) to add
methods that return a SSLCertChain directly, since it results in a
somewhat cleaner object model.
So this CL switches everything to use the "chain" methods, and gets
rid of the obsolete methods and member variables.
Bug: webrtc:8920
Change-Id: Ie9d7d53654ba859535462521b54c788adec7badf
Reviewed-on: https://webrtc-review.googlesource.com/56961
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22177}
JsepTransportController process the entire SDP and handle the RTCP-mux,
SRTP setup, BUNDLE related logic internally. This will replace the current
TransportController.
JsepTransport2 is used by the JsepTransportController which processes the
transport part of SDP and owns the DtlsTransport created internally.
JsepTransport2 will replace JsepTransport and be renamed eventually.
Bug: webrtc:8587
Change-Id: Ib02dfa52fe9b7a5b8b132afcc8e4363eb8bd9cf4
Reviewed-on: https://webrtc-review.googlesource.com/48841
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22164}
STUN candidates use STUN binding requests to keep NAT bindings open.
Related stats including packet loss and RTT can be now collected via the
legacy GetStats in PeerConnection.
Bug: None
Change-Id: I7b0eee1ccb07eb670a32ee303c9590047b25f31c
Reviewed-on: https://webrtc-review.googlesource.com/54100
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22113}
STUN candidates use STUN binding requests to keep NAT bindings open. The
interval at which the STUN keepalive pings are sent is configurable now
via RTCConfiguration.
TBR=sakal@webrtc.org
Bug: None
Change-Id: I5f99ea3fe1e9042fa2bf7dcab0aace78f57739e6
Reviewed-on: https://webrtc-review.googlesource.com/54180
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22109}
This reverts commit 6780c51b23516803dc27173d10ba98d018780447.
Reason for revert:
More details in crbug.com/810292
Original change's description:
> Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32.
>
> A field has been added to "CryptoOptions" to enable SRTP_AES128_CM_SHA1_32
> from native apps if really necessary.
>
> R=deadbeef@webrtc.org
>
> Bug: webrtc:7670
> Change-Id: I36b6ab3e302fbf3cda2611ff196757e43a56e704
> Reviewed-on: https://webrtc-review.googlesource.com/41420
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21952}
TBR=deadbeef@webrtc.org,magjed@webrtc.org,jbauch@webrtc.org
Change-Id: I643dbe023eca526f2cda4d97df045f2533741dd4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7670
Reviewed-on: https://webrtc-review.googlesource.com/49880
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21961}
The bug 8432 is caused by trying to connect through a
"link-local" interface (IP address 169.254.0.x/16),
which is listed among the iPhone network interfaces.
The bug is not happening if the link-local network interfaces
are skipped in the ICE candidate gethering process.
To control this behaviour an option - disable_link_local_networks -
is added inside the RTCConfiguration.
It is used to set the new BasicPortAllocatorSession flag -
PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS.
The port allocator flag is added if the configuration option is set.
IPIsLinkLocal IPAddress function and its friends (IPIsLoopback, IPIsPrivate)
are refactored to work on both IPv4 and IPv6.
Unit test IPIsLinkLocal.
Bonus: fix a bug in IPIsLinkLocalV6:
take into account just 10 network mask bits instead of 16.
Bug: webrtc:8432
Change-Id: Ibe8f677a36098057b7fcad5c798380727b23359b
Reviewed-on: https://webrtc-review.googlesource.com/36380
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21922}
This patch modifies behaviour when TurnPort gets a lifetime
back from server that is shorter than 2 minutes.
Before the patch such lifetime resulted in TurnPort not scheduling any
refresh, leading to timeout on the turn allocation.
After then patch lifetime shorter then 2 minutes leads to refresh
after half stipulated lifetime.
BUG=webrtc:8826
Change-Id: I80561100f2307bd9a6a91af0924bb2814102ddd3
Reviewed-on: https://webrtc-review.googlesource.com/46741
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21891}
This change list passes the instance of RtcEventLog from Peerconnection
down to P2PTransportChannel, and binds the structured ICE logging with
ICE layer objects. Logs of ICE connectivity checks are injected for
candidate pairs.
TBR=terelius@webrtc.org
Bug: None
Change-Id: Ia979dbbac6d31dcf0f8988da1065bdfc3e461821
Reviewed-on: https://webrtc-review.googlesource.com/34660
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21884}
The network preference is added to RTCConfiguration and passed to ICE.
ICE considers now the preference set by applications over network
interface types when making decisions in candidate pair switching.
Bug: webrtc:8816
Change-Id: I40d2612705b54c83dd45772ac855808e0a76b1e1
Reviewed-on: https://webrtc-review.googlesource.com/44020
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21855}
This method returns the DTLS SSL certificate chain associated with the
audio transport on the remote side. This will become populated once the
DTLS connection with the peer has been completed.
TBR=deadbeef@webrtc.org
Bug: webrtc:8800
Change-Id: Ib90ccb3463415e798c17c187c5bdbfc4da26f11f
Reviewed-on: https://webrtc-review.googlesource.com/44140
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21785}
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.
Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).
This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.
Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.
TBR=phoglund@webrtc.org
Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
On Chrome Remote Desktop for iOS, if all of these are true:
* Enable the PORTALLOCATOR_DISABLE_COSTLY_NETWORKS flag.
* Connect an iPhone to a Mac.
* Turn off WiFi on the iPhone and keep mobile data on.
* Connect to a host on a different network from iPhone.
Then the connection can never succeed. The reason is that iOS uses a
special network interface with link-local IP to communicate with the Mac.
BasicPortAllocator sees that interface and thinks it costs less than the
cellular networks, then it removes all cellular networks. However, that
link-local network cannot connect to a peer on external network.
This CL changes the behavior of the PORTALLOCATOR_DISABLE_COSTLY_NETWORKS
so that it ignores the cost of link-local networks when determining the
lowest network cost.
Bug: webrtc:8780
Change-Id: I9bde50426b356fdbf89d4b826fc7b28c8c523c10
Reviewed-on: https://webrtc-review.googlesource.com/41460
Commit-Queue: Yuwei Huang <yuweih@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21738}
This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613.
Reason for revert:
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660
Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
>
> This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
>
> Reason for revert:
>
> Breaks Chrome FYI:
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
> -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
> static_library(target_name) {
> ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
> //third_party/webrtc/*
> //third_party/webrtc_overrides/*
> ]
>
> https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
>
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> >
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> >
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> >
> > BUG=webrtc:8254
> >
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
>
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
Reason for revert:
Breaks Chrome FYI:
/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
static_library(target_name) {
^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
//third_party/webrtc/*
//third_party/webrtc_overrides/*
]
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
>
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
>
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
>
> BUG=webrtc:8254
>
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.
API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.
BUG=webrtc:8254
Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
Now that downstream projects have been updated to point to
pc/sessiondescription.h, the forwarding header can be removed.
Bug: webrtc:8620
Change-Id: Ia4b4aa05f41a2b6ef948dc140460b71d8db8eb64
Reviewed-on: https://webrtc-review.googlesource.com/35961
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21418}
This makes the following changes:
- Replaces ContentDescription with its only subclass,
MediaContentDescription
- Adds helpers to cast a MediaContentDescription to its
audio, video, and data subclasses.
- Changes ContentInfo.type to a new enum, MediaProtocolType.
Bug: webrtc:8620
Change-Id: I5eb0811cb16a51b0b9d73ecc4fe8edc7037f1aed
Reviewed-on: https://webrtc-review.googlesource.com/35100
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21401}
Was just being kept around until downstream dependents were updated
to include other files.
BUG=None
Change-Id: I26f0bea83c026be8a50a259c229ef8e7445825ad
Reviewed-on: https://webrtc-review.googlesource.com/35009
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21395}
SDP is a detail of PeerConnection and is not used by anything in p2p, so
it belongs in the pc/ directory. This also allows
MediaContentDescription to be inlined in the future.
Bug: webrtc:8620
Change-Id: I38b65ede9942e29eb15035ab29f2be988da1e5ce
Reviewed-on: https://webrtc-review.googlesource.com/33781
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21376}
This is a reland of 4770fd935ac92400487bddd3b755753572e6d692
Original change's description:
> Move JsepTransport from p2p/base to pc/.
>
> The JsepTransport class is moved to pc/ and the utility methods and
> enums are moved to where they are used.
>
> With JsepTransport moved to pc/, JsepTransport can depend on objects in
> pc/ including RtpTranport, SrtpTransport etc.
>
> Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7
>
> Bug: webrtc:8636
> Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
> Reviewed-on: https://webrtc-review.googlesource.com/33701
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21333}
Bug: webrtc:8636
Change-Id: Ibce42be898b96dd8e0266b595611d2ffc86581a8
Reviewed-on: https://webrtc-review.googlesource.com/34586
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21371}
https://webrtc.googlesource.com/src.git/+/26246cac660a95f439b7d1c593edec2929806d3f
that was reverted due to compile error on windows.
Changes since last is an addition of a cast to uint16_t in stun.cc:1018.
---
Add RelayPortFactoryInterface that allows for custom relay (e.g turn) ports
This patch adds a RelayPortFactoryInterface that allows
for custom relay ports. The factor is added as optional argument
to BasicPortAlloctor. If none is provided a default implementation
that mimics existing behavior is created.
The patch also adds 2 stun functions, namely to copy a
StunAttribute and to remove StunAttribute's from a StunMessage.
Bug: webrtc:8640
Change-Id: If23638317130060286f576c94401de55c60a1821
Reviewed-on: https://webrtc-review.googlesource.com/34181
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21345}
This reverts commit 4770fd935ac92400487bddd3b755753572e6d692.
Reason for revert: breaks downstream projects
Original change's description:
> Move JsepTransport from p2p/base to pc/.
>
> The JsepTransport class is moved to pc/ and the utility methods and
> enums are moved to where they are used.
>
> With JsepTransport moved to pc/, JsepTransport can depend on objects in
> pc/ including RtpTranport, SrtpTransport etc.
>
> Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7
>
> Bug: webrtc:8636
> Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
> Reviewed-on: https://webrtc-review.googlesource.com/33701
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21333}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,pthatcher@webrtc.org
Change-Id: Ia72c6d7f185a95b21fd0aec90e7fdc00cb1fb423
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8636
Reviewed-on: https://webrtc-review.googlesource.com/34600
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21335}
