This change adds exposure of a new transceiver method for getting
the total set of supported extensions stored as an attribute,
and their direction. If the direction is kStopped, the extension
is not signalled in Unified Plan SDP negotiation.
Note: SDP negotiation is not modified by this change.
Changes:
- RtpHeaderExtensionCapability gets a new RtpTransceiverDirection,
indicating either kStopped (extension available but not signalled),
or other (extension signalled).
- RtpTransceiver gets the new method as described above. The
default value of the attribute comes from the voice and video
engines as before.
https://chromestatus.com/feature/5680189201711104.
go/rtp-header-extension-ip
Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk
Bug: chromium:1051821
Change-Id: I440443b474db5b1cfe8c6b25b6c10a3ff9c21a8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170235
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30800}
Split the sctp_mid_ variable into two variables,
sctp_mid_n_ and sctp_mid_s_, each of which is only accessed
by one thread.
Bug: webrtc:9987
Change-Id: I4dce944b920f4698e2606a7b85776791cbf55c28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168243
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30503}
This was an ICE configuration experiment added a couple years ago that did not end up being used.
Bug: webrtc:11316
Change-Id: Iafb7e1c4f7b4598815f045808dbf6e470172f119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167680
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30395}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
from LS_INFO to LS_VERBOSE.
By default, unit tests run with logging at info level.
A random run today produced more than 70.000 lines of
output. This CL would reduce that by approximately 15.000.
Bug: none
Change-Id: Ie62708cebf109510a2443aa5ab5c4e645ffc6707
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161950
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30077}
I could find no reason for the extra complexity of doing messaging
in order to schedule a task to be done after the current cycle.
It also simplifies the peerconnection/datachannelcontroller coupling.
Bug: webrtc:11146
Change-Id: I68f45059b9f4a6869fb44b856e05a480f4652365
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161232
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29997}
This CL:
- Moved HasDataChannel and data_channel_type_
- Moved rtp_data_channels_
- Moved sctp_data_channels_
- Moved data_channel_controller to its own .h file
- Various changes to reduce the coupling between the classes
- Removed friendship between DataChannelController and PeerConnection
Bug: webrtc:11146
Change-Id: Ib8c395e4c90ce34baf40812d1dade0ffa79f2438
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161094
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29987}
This is step 1-3 of the refactoring process outlined in comment #1 of bugs.webrtc.org/11146
Bug: webrtc:11146
Change-Id: Iccad009bc0585f99d207a6ddb42fd8e71312fc0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161003
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29970}
MediaTransport is deprecated and the code is unused.
No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
This is the same as the existing version, except it uses the Operations
Chain. As such, if an asynchronous operation that uses the chain is
currently pending, such as CreateOffer() or CreateAnswer(),
AddIceCandidate() will not happen until the previous operation
completes.
Bug: chromium:1019222
Change-Id: Ie6e5fc386fa9c29b5e2f8e3f65bfbaf9837d351c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158741
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29704}
This is a reland of 1dddaa1a84330091ca083c950ef2e24a85a48fc8
The regression that caused the original CL to be reverted was the fact that
invoking SetLocalDescription() inside of the CreateOffer() callback was no
longer executing synchronously and immediately.
In this CL, the original CL is patched so that the CreateOffer() operation
is marked as completed just before invoking the CreateOffer() callback
(versus doing it just afterwards). This ensures that the OperationsChain is
popped before the callback runs. The same applies for CreateAnswer().
See diff between Patch Set 1 (Original CL) and the latest Patch Set.
Original change's description:
> [PeerConnection] Use an OperationsChain in PeerConnection for async ops.
>
> For background, motivation, requirements and implementation notes, see
> https://docs.google.com/document/d/1XLwNN2kUIGGTwz9LQ0NwJNkcybi9oKnynUEZB1jGA14/edit?usp=sharing
>
> Using the OperationsChain will unblock future CLs from chaining multiple
> operations together such as implementing parameterless
> setLocalDescription().
>
> In this CL, the OperationsChain is used in existing signaling operations
> with little intended side-effects. An operation that is chained onto an
> empty OperationsChain will for instance execute immediately, and
> SetLocalDescription() and SetRemoteDescription() are implemented as
> "synchronous operations".
>
> The lifetime of the PeerConnection is not indended to change as a result
> of this CL: All chained operations use a WeakPtr to the PC to ensure
> use-after-free does not happen.
>
> There is one notable change though: CreateOffer() and CreateAnswer() will
> asynchronously delay other signaling methods from executing until they
> have completed.
>
> Drive-by fix: This CL also ensures that early failing
> CreateOffer/CreateAnswer operation's observers are invoked if the
> PeerConnection is destroyed while a PostCreateSessionDescriptionFailure
> is pending.
>
> Bug: webrtc:11019
> Change-Id: I521333e41d20d9bbfb1e721609f2c9db2a5f93a9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157305
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29605}
TBR=steveanton@webrtc.org
Bug: webrtc:11019
Change-Id: I57b4496e63378c91c24679ee496e21f5cb6a0e59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158524
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29630}
This reverts commit 1dddaa1a84330091ca083c950ef2e24a85a48fc8.
Reason for revert: Breaks downstream projects :(
Original change's description:
> [PeerConnection] Use an OperationsChain in PeerConnection for async ops.
>
> For background, motivation, requirements and implementation notes, see
> https://docs.google.com/document/d/1XLwNN2kUIGGTwz9LQ0NwJNkcybi9oKnynUEZB1jGA14/edit?usp=sharing
>
> Using the OperationsChain will unblock future CLs from chaining multiple
> operations together such as implementing parameterless
> setLocalDescription().
>
> In this CL, the OperationsChain is used in existing signaling operations
> with little intended side-effects. An operation that is chained onto an
> empty OperationsChain will for instance execute immediately, and
> SetLocalDescription() and SetRemoteDescription() are implemented as
> "synchronous operations".
>
> The lifetime of the PeerConnection is not indended to change as a result
> of this CL: All chained operations use a raw pointer to the PC that is
> ensured not to be used-after-free using an "IsAlive" object.
>
> There is one notable change though: CreateOffer() and CreateAnswer() will
> asynchronously delay other signaling methods from executing until they
> have completed.
>
> Drive-by fix: This CL also ensures that early failing
> CreateOffer/CreateAnswer operation's observers are invoked if the
> PeerConnection is destroyed while a PostCreateSessionDescriptionFailure
> is pending.
>
> Bug: webrtc:11019
> Change-Id: I521333e41d20d9bbfb1e721609f2c9db2a5f93a9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157305
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29605}
TBR=steveanton@webrtc.org,hbos@webrtc.org
Change-Id: Ie540dcc8ecdc48ad0c65d23645fbc3ad5f99592b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11019
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158405
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29611}
For background, motivation, requirements and implementation notes, see
https://docs.google.com/document/d/1XLwNN2kUIGGTwz9LQ0NwJNkcybi9oKnynUEZB1jGA14/edit?usp=sharing
Using the OperationsChain will unblock future CLs from chaining multiple
operations together such as implementing parameterless
setLocalDescription().
In this CL, the OperationsChain is used in existing signaling operations
with little intended side-effects. An operation that is chained onto an
empty OperationsChain will for instance execute immediately, and
SetLocalDescription() and SetRemoteDescription() are implemented as
"synchronous operations".
The lifetime of the PeerConnection is not indended to change as a result
of this CL: All chained operations use a raw pointer to the PC that is
ensured not to be used-after-free using an "IsAlive" object.
There is one notable change though: CreateOffer() and CreateAnswer() will
asynchronously delay other signaling methods from executing until they
have completed.
Drive-by fix: This CL also ensures that early failing
CreateOffer/CreateAnswer operation's observers are invoked if the
PeerConnection is destroyed while a PostCreateSessionDescriptionFailure
is pending.
Bug: webrtc:11019
Change-Id: I521333e41d20d9bbfb1e721609f2c9db2a5f93a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157305
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29605}
This is a reland of 16d4c4d4fbb8644033def1091d2d5c941c1b01fa after
downstream project was updated to be prepared for the new SdpType.
Original change's description:
> Implement rollback for setRemoteDescription
>
> Bug: chromium:980875
> Change-Id: I4575e9ad1902a20937f9812f49edee2a2441f76d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153525
> Commit-Queue: Eldar Rello <elrello@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29422}
TBR=steveanton@webrtc.org
Bug: chromium:980875
Change-Id: Iba8d25bf2dc481b25a03eeae9818bd5f4c3eaa2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156569
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29460}
Bug: chromium:980875
Change-Id: I4575e9ad1902a20937f9812f49edee2a2441f76d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153525
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29422}
The plugin transport parameters (a=x-opaque: lines) relate to how to create and
set up a plugin transport. When SDP bundle is used, the x-opaque line needs to
be copied into the bundled m= section. This means x-opaque can appear on a
section even if the offerer does not intend to use the transport for the media
described by that section. Consequently, the answerer cannot currently tell
whether the caller is offering an alternate transport for media, data, or both.
This change adds an a=x-alt-protocol: line to SDP. The value following this
line matches the <protocol> part of the x-opaque:<protocol>:<params> line.
However, alt-protocol is not bundled--it only ever applies to the m= section
that contains the line. This allows the offerer to express which m= sections
should actually use an alternate transport, even in the case of bundle.
Note that this is still limited by the available configuration options:
datagram transport can be used for media (audio + video) and/or data. It is
still not possible to use it for audio but not video, or vice versa.
PeerConnection places an alt-protocol line in each media (audio/video) m=
section if it is configured to use a datagram transport for media. It places
an alt-protocol line in each data m= section if it is configured to use a
datagram transport for data channels. PeerConnection leaves alt-protocol in
media (audio/video) m= sections of the answer if it is configured to use a
datagram transport for media, and in data m= sections of the answer if it is
configured to use a datagram transport for data channels.
JsepTransport now negotiates use of the datagram transport independently for
media and data channels. It only uses it for media if the m= sections for
bundled audio/video have an alt-protocol line matching the x-opaque protocol,
and only uses it for data channels if a bundled m= section for data has an
alt-protocol line matching the x-opaque protocol.
Bug: webrtc:9719
Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29351}
Adds a field trial and configuration parameter to control whether
datagram transport may be used for data channels in a receive-only
manner. By default, if use_datagram_transport_for_data_channels is
enabled, PeerConnection will create a datagram transport and offer its
use for outgoing calls as well as accept incoming offers with compatible
datagram transport parameters.
With this change, a receive_only mode is added for datagram transport
data channels. When receive_only is set, the PeerConnection will not
create or offer datagram transports for outgoing calls, but will accept
incoming calls that offer compatible datagram transport parameters.
Bug: webrtc:9719
Change-Id: I35667bcc408ea4bbc61155898e6d2472dd262711
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154463
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29327}
This is a reland of 487f9a17e426fd14bb06b13e861071b3f15d119b
Original change's description:
> Reland "Refactor SCTP data channels to use DataChannelTransportInterface."
>
> Also clears SctpTransport before deleting JsepTransport.
>
> SctpTransport is ref-counted, but the underlying transport is deleted when
> JsepTransport clears the rtp_dtls_transport. This results in crashes when
> usrsctp attempts to send outgoing packets through a dangling pointer to the
> underlying transport.
>
> Clearing SctpTransport before DtlsTransport removes the pointer to the
> underlying transport before it becomes invalid.
>
> This fixes a crash in chromium's web platform tests (see
> https://chromium-review.googlesource.com/c/chromium/src/+/1776711).
>
> Original change's description:
> > Refactor SCTP data channels to use DataChannelTransportInterface.
> >
> > This change moves SctpTransport to be owned by JsepTransport, which now
> > holds a DataChannelTransport implementation for SCTP when it is used for
> > data channels.
> >
> > This simplifies negotiation and fallback to SCTP. Negotiation can now
> > use a composite DataChannelTransport, just as negotiation for RTP uses a
> > composite RTP transport.
> >
> > PeerConnection also has one fewer way it needs to manage data channels.
> > It now handles SCTP and datagram- or media-transport-based data channels
> > the same way.
> >
> > There are a few leaky abstractions left. For example, PeerConnection
> > calls Start() on the SctpTransport at a particular point in negotiation,
> > but does not need to call this for other transports. Similarly, PC
> > exposes an interface to the SCTP transport directly to the user; there
> > is no equivalent for other transports.
>
> Bug: webrtc:9719
> Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29120}
Bug: webrtc:9719
Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29290}
PeerConnection now watches when data channels become ready to send
through its implementation of DataChannelSink, and no longer needs to
monitor the MediaTransport state.
Bug: webrtc:9719
Change-Id: I3e17747eb03926a3791c204bf5a1d2dc67855c09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154001
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29261}
Previously, each RTCP packet was handled several times in a row, once
per m-section. This caused various weirdness and log warning spam, in
particular when using unified plan.
The cause was that the packets were wired trough each BaseChannel
instance up to the Call class. With this fix, the RTCP packets are wired
once per RtpTransportInternal via the common peer connection class.
Bug: chromium:1002875
Change-Id: I41c4eb3b68e215ebe0f2c6fb93ae0ee73335b89a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152668
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29226}
Changes Rtp Receivers to use a null value of ssrc to mean a default
receive stream.
Bug: webrtc:8694
Change-Id: I835199345f7add993b9078c8b0e7988d5cdd6646
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152425
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29201}
This reverts commit 487f9a17e426fd14bb06b13e861071b3f15d119b.
Reason for revert: speculative revert
Original change's description:
> Reland "Refactor SCTP data channels to use DataChannelTransportInterface."
>
> Also clears SctpTransport before deleting JsepTransport.
>
> SctpTransport is ref-counted, but the underlying transport is deleted when
> JsepTransport clears the rtp_dtls_transport. This results in crashes when
> usrsctp attempts to send outgoing packets through a dangling pointer to the
> underlying transport.
>
> Clearing SctpTransport before DtlsTransport removes the pointer to the
> underlying transport before it becomes invalid.
>
> This fixes a crash in chromium's web platform tests (see
> https://chromium-review.googlesource.com/c/chromium/src/+/1776711).
>
> Original change's description:
> > Refactor SCTP data channels to use DataChannelTransportInterface.
> >
> > This change moves SctpTransport to be owned by JsepTransport, which now
> > holds a DataChannelTransport implementation for SCTP when it is used for
> > data channels.
> >
> > This simplifies negotiation and fallback to SCTP. Negotiation can now
> > use a composite DataChannelTransport, just as negotiation for RTP uses a
> > composite RTP transport.
> >
> > PeerConnection also has one fewer way it needs to manage data channels.
> > It now handles SCTP and datagram- or media-transport-based data channels
> > the same way.
> >
> > There are a few leaky abstractions left. For example, PeerConnection
> > calls Start() on the SctpTransport at a particular point in negotiation,
> > but does not need to call this for other transports. Similarly, PC
> > exposes an interface to the SCTP transport directly to the user; there
> > is no equivalent for other transports.
>
> Bug: webrtc:9719
> Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29120}
TBR=steveanton@webrtc.org,mellem@webrtc.org,benwright@webrtc.org
Change-Id: Ibd1a7f30931c114212c90824fec414d276d3f915
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152421
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29141}
Also clears SctpTransport before deleting JsepTransport.
SctpTransport is ref-counted, but the underlying transport is deleted when
JsepTransport clears the rtp_dtls_transport. This results in crashes when
usrsctp attempts to send outgoing packets through a dangling pointer to the
underlying transport.
Clearing SctpTransport before DtlsTransport removes the pointer to the
underlying transport before it becomes invalid.
This fixes a crash in chromium's web platform tests (see
https://chromium-review.googlesource.com/c/chromium/src/+/1776711).
Original change's description:
> Refactor SCTP data channels to use DataChannelTransportInterface.
>
> This change moves SctpTransport to be owned by JsepTransport, which now
> holds a DataChannelTransport implementation for SCTP when it is used for
> data channels.
>
> This simplifies negotiation and fallback to SCTP. Negotiation can now
> use a composite DataChannelTransport, just as negotiation for RTP uses a
> composite RTP transport.
>
> PeerConnection also has one fewer way it needs to manage data channels.
> It now handles SCTP and datagram- or media-transport-based data channels
> the same way.
>
> There are a few leaky abstractions left. For example, PeerConnection
> calls Start() on the SctpTransport at a particular point in negotiation,
> but does not need to call this for other transports. Similarly, PC
> exposes an interface to the SCTP transport directly to the user; there
> is no equivalent for other transports.
Bug: webrtc:9719
Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29120}
This is a canary CL to check if using c++14 feature breaks any webrtc user.
Bug: webrtc:10945
Change-Id: Iabaf8c06414c1ac960791bcb7cc46f5f5a5e1f14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151600
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29119}
This reverts commit 4c85828ab272d9bd58789bad7b135b6287395f97.
Reason for revert:
Speculatively reverting this because it makes several web platform tests relating to RTCDataChannel flaky, see first failing roll:
https://chromium-review.googlesource.com/c/chromium/src/+/1776711
Original change's description:
> Refactor SCTP data channels to use DataChannelTransportInterface.
>
> This change moves SctpTransport to be owned by JsepTransport, which now
> holds a DataChannelTransport implementation for SCTP when it is used for
> data channels.
>
> This simplifies negotiation and fallback to SCTP. Negotiation can now
> use a composite DataChannelTransport, just as negotiation for RTP uses a
> composite RTP transport.
>
> PeerConnection also has one fewer way it needs to manage data channels.
> It now handles SCTP and datagram- or media-transport-based data channels
> the same way.
>
> There are a few leaky abstractions left. For example, PeerConnection
> calls Start() on the SctpTransport at a particular point in negotiation,
> but does not need to call this for other transports. Similarly, PC
> exposes an interface to the SCTP transport directly to the user; there
> is no equivalent for other transports.
>
> Bug: webrtc:9719
> Change-Id: I0d3151c48c1a511368277981fc4cf818a9f8ebb4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150341
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29012}
TBR=steveanton@webrtc.org,mellem@webrtc.org,benwright@webrtc.org
Change-Id: I074b9e68f298d20d0cabb4239084b4843e76e910
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150944
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29025}
This change moves SctpTransport to be owned by JsepTransport, which now
holds a DataChannelTransport implementation for SCTP when it is used for
data channels.
This simplifies negotiation and fallback to SCTP. Negotiation can now
use a composite DataChannelTransport, just as negotiation for RTP uses a
composite RTP transport.
PeerConnection also has one fewer way it needs to manage data channels.
It now handles SCTP and datagram- or media-transport-based data channels
the same way.
There are a few leaky abstractions left. For example, PeerConnection
calls Start() on the SctpTransport at a particular point in negotiation,
but does not need to call this for other transports. Similarly, PC
exposes an interface to the SCTP transport directly to the user; there
is no equivalent for other transports.
Bug: webrtc:9719
Change-Id: I0d3151c48c1a511368277981fc4cf818a9f8ebb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150341
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29012}
This patch adds accounting and reporting needed for
newly added RTCTransportStats.selectedCandidatePairChanges,
https://w3c.github.io/webrtc-stats/#dom-rtctransportstats-selectedcandidatepairchanges
a) P2PTransportChannel counts everytime selected_connection_
is modified and reports this counter in the GetStats()-call.
b) RTCStatsCollector puts the counter into the standardized
stats object.
Bug: webrtc:10900
Change-Id: Ibaeca18706b8edcbcb44b0c6f2754854bcb545ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149830
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28987}
This patch adds a new (optional) attribute to TURN_ALLOCATE_REQUEST,
TURN_LOGGING_ID (0xFF05).
The attribute is put into the comprehension-optional range
so that a TURN server should ignore it if it doesn't know if.
https://tools.ietf.org/html/rfc5389#section-18.2
The intended usage of this attribute is to correlate client and
backend logs.
Bug: webrtc:10897
Change-Id: I51fdbe15f9025e817cd91ee8e2c3355133212daa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149829
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28966}
PeerConnection now has a new setting in RTCConfiguration to enable use of
datagram transport for data channels. There is also a corresponding field
trial, which has both a kill-switch and a way to change the default value.
PeerConnection's interaction with MediaTransport for data channels has been
refactored to work with DataChannelTransportInterface instead.
Adds a DataChannelState and OnStateChanged() to the DataChannelSink
callbacks. This allows PeerConnection to listen to the data channel's
state directly, instead of indirectly by monitoring media transport
state. This is necessary to enable use of non-media-transport (eg.
datagram transport) data channel transports.
For now, PeerConnection watches the state through MediaTransport as well.
This will persist until MediaTransport implements the new callback.
Datagram transport use is negotiated. As such, an offer that requests to use
datagram transport for data channels may be rejected by the answerer. If the
offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data
channels with an extra x-opaque parameter for datagram transport. If the
opaque parameter is rejected (by an answerer without datagram support), the
offerer may fall back to SCTP.
If DTLS is not enabled, there is no viable fallback. In this case, the data
channels are negotiated as media transport data channels. If the receiver does
not understand the x-opaque line, it will reject these data channels, and the
offerer's data channels will be closed.
Bug: webrtc:9719
Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28932}
