Datagram_dtls_adaptor needs access to rtp_rtcp modules and this moves helps to keep p2p/base/ without dependency on rtp_rtcp.
Bug: webrtc:9719
Change-Id: Ic337be3fb9f68106187a84efa815eefbe5b0fcd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145267
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28533}
outgoing checks.
This change adds an experimental feature to allow an ICE agent to embed
the transaction ID of the latest connectivity check received from the
remote peer, as an auxiliary acknowledgement in additional to the check
response, in its own checks. This could facilitate the establishment of
ICE connectivity if the check process has a high RTT.
Bug: None
Change-Id: If3e6327720f13beeb14f103af3b5ffb4f9692998
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142682
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28316}
In short, the caller places a x-opaque line in SDP for each m= section that
uses datagram transport. If the answerer supports datagram transport, it will
parse this line and create a datagram transport. It will then echo the x-opaque
line into the answer (to indicate that it accepted use of datagram transport).
If the offer and answer contain exactly the same x-opaque line, both peers will
use datagram transport. If the x-opaque line is omitted from the answer (or is
different in the answer) they will fall back to RTP.
Note that a different x-opaque line in the answer means the answerer did not
understand something in the negotiation proto. Since WebRTC cannot know what
was misunderstood, or whether it's still possible to use the datagram transport,
it must fall back to RTP. This may change in the future, possibly by passing
the answer to the datagram transport, but it's good enough for now.
Negotiation consists of four parts:
1. DatagramTransport exposes transport parameters for both client and server
perspectives. The client just echoes what it received from the server (modulo
any fields it might not have understood).
2. SDP adds a x-opaque line for opaque transport parameters. Identical to
x-mt, but this is specific to datagram transport and goes in each m= section,
and appears in the answer as well as the offer.
- This is propagated to Jsep as part of the TransportDescription.
- SDP files: transport_description.h,cc, transport_description_factory.h,cc,
media_session.cc, webrtc_sdp.cc
3. JsepTransport/Controller:
- Exposes opaque parameters for each mid (m= section). On offerer, this means
pre-allocating a datagram transport and getting its parameters. On the
answerer, this means echoing the offerer's parameters.
- Uses a composite RTP transport to receive from either default RTP or
datagram transport until both offer and answer arrive.
- If a provisional answer arrives, sets the composite to send on the
provisionally selected transport.
- Once both offer and answer are set, deletes the unneeded transports and
keeps whichever transport is selected.
4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
Bug: webrtc:9719
Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28198}
This reverts commit 71c6482baf0ff17141c635e6a7639493db68a65c.
Reason for revert: Lands too much at once and breaks downstream tests that need to implement new interfaces first.
Original change's description:
> Implement true negotiation for DatagramTransport with fallback to RTP.
>
> In short, the caller places a x-opaque line in SDP for each m= section that
> uses datagram transport. If the answerer supports datagram transport, it will
> parse this line and create a datagram transport. It will then echo the x-opaque
> line into the answer (to indicate that it accepted use of datagram transport).
>
> If the offer and answer contain exactly the same x-opaque line, both peers will
> use datagram transport. If the x-opaque line is omitted from the answer (or is
> different in the answer) they will fall back to RTP.
>
> Note that a different x-opaque line in the answer means the answerer did not
> understand something in the negotiation proto. Since WebRTC cannot know what
> was misunderstood, or whether it's still possible to use the datagram transport,
> it must fall back to RTP. This may change in the future, possibly by passing
> the answer to the datagram transport, but it's good enough for now.
>
> Negotiation consists of four parts:
> 1. DatagramTransport exposes transport parameters for both client and server
> perspectives. The client just echoes what it received from the server (modulo
> any fields it might not have understood).
>
> 2. SDP adds a x-opaque line for opaque transport parameters. Identical to
> x-mt, but this is specific to datagram transport and goes in each m= section,
> and appears in the answer as well as the offer.
> - This is propagated to Jsep as part of the TransportDescription.
> - SDP files: transport_description.h,cc, transport_description_factory.h,cc,
> media_session.cc, webrtc_sdp.cc
>
> 3. JsepTransport/Controller:
> - Exposes opaque parameters for each mid (m= section). On offerer, this means
> pre-allocating a datagram transport and getting its parameters. On the
> answerer, this means echoing the offerer's parameters.
> - Uses a composite RTP transport to receive from either default RTP or
> datagram transport until both offer and answer arrive.
> - If a provisional answer arrives, sets the composite to send on the
> provisionally selected transport.
> - Once both offer and answer are set, deletes the unneeded transports and
> keeps whichever transport is selected.
>
> 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
>
> Bug: webrtc:9719
> Change-Id: Id8996eb1871e79d93b7923a5d7eb3431548c798d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140700
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28182}
TBR=steveanton@webrtc.org,mellem@webrtc.org,sukhanov@webrtc.org
Change-Id: I0d502c4a6d27516c35ed85154f3fa5869f88b3b7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140822
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28188}
In short, the caller places a x-opaque line in SDP for each m= section that
uses datagram transport. If the answerer supports datagram transport, it will
parse this line and create a datagram transport. It will then echo the x-opaque
line into the answer (to indicate that it accepted use of datagram transport).
If the offer and answer contain exactly the same x-opaque line, both peers will
use datagram transport. If the x-opaque line is omitted from the answer (or is
different in the answer) they will fall back to RTP.
Note that a different x-opaque line in the answer means the answerer did not
understand something in the negotiation proto. Since WebRTC cannot know what
was misunderstood, or whether it's still possible to use the datagram transport,
it must fall back to RTP. This may change in the future, possibly by passing
the answer to the datagram transport, but it's good enough for now.
Negotiation consists of four parts:
1. DatagramTransport exposes transport parameters for both client and server
perspectives. The client just echoes what it received from the server (modulo
any fields it might not have understood).
2. SDP adds a x-opaque line for opaque transport parameters. Identical to
x-mt, but this is specific to datagram transport and goes in each m= section,
and appears in the answer as well as the offer.
- This is propagated to Jsep as part of the TransportDescription.
- SDP files: transport_description.h,cc, transport_description_factory.h,cc,
media_session.cc, webrtc_sdp.cc
3. JsepTransport/Controller:
- Exposes opaque parameters for each mid (m= section). On offerer, this means
pre-allocating a datagram transport and getting its parameters. On the
answerer, this means echoing the offerer's parameters.
- Uses a composite RTP transport to receive from either default RTP or
datagram transport until both offer and answer arrive.
- If a provisional answer arrives, sets the composite to send on the
provisionally selected transport.
- Once both offer and answer are set, deletes the unneeded transports and
keeps whichever transport is selected.
4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
Bug: webrtc:9719
Change-Id: Id8996eb1871e79d93b7923a5d7eb3431548c798d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140700
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28182}
- This makes it consistent with ICE and MediaTransport ownership.
- Removes unnecessary datagram_transport() getter in DtlsTransportInternal
As a side effect this fixes bug in JsepTransportController, which moved datagram_transport to Dtls after creating it, then checked if (datagram_transport) to decide which RTP transport to create. As a result of this bug we were creating Sded instead of Unencrypted RTP with datagram transport.
Bug: webrtc:9719
Change-Id: Ic5b13a450ce6ac5b2a20d388657e3949aabef079
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139620
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28146}
It does not make sense for DtlsTransport to own ICE, and this arrangement will
not work when negotiating datagram or DTLS transport. During negotiation, both
a DTLS transport and a datagram transport need to be ready to receive from the
same ICE transport, depending on which protocol is chosen by the answerer. Once
the answerer chooses a protocol, the transport that is not chosen must be
deleted, but ICE must be left intact for use by the remaining transport.
Bug: webrtc:9719
Change-Id: Ibab969b574c981e3834ced71f8ff88008cb26a6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139340
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28113}
This patch is a NOP and moves
- class Connection
- class ConnectionInfo
- class ProxyConnection
from port.{h/cc} to a new file called connection.{h/cc}
BUG=webrtc:10647
Change-Id: I89322d3421d272657e24a46b28ab6679fcdc9450
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137509
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28101}
Connect ICE state changes to datagram transport regardless of bypass mode.
ICE states were connected to datagram transport only in bypass mode. As a result, if we received datagram state change notification before ICE state change notification, the state was not propagated.
TODO: We need fake datagram transport implementation/test so that we could unit test such failures without relying on downstream projects.
Bug: webrtc:9719
Change-Id: I5a180676e0d05f707b2a43d07e8c04fb10985027
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138982
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28094}
This will be used to multiplex multiple transports during SDP
negotiation. When the offerer watns to support multiple RTP transports,
it will combine them into a singla CompositeRtpTransport.
CompositeRtpTransport can receive from any of the offered transports
while waiting for an answer to arrive.
The choice of which transport is used to send must be driven by the SDP
answer. If a provisional answer arrives, the composite can be set to
send using the chosen transport, while maintaining other transports in
case the peer changes its mind. When the final answer arrives, the
composite will be deleted and replaced with the chosen transport.
Bug: webrtc:9719
Change-Id: Ib8cea77ef202f37086723bfa2c71e2aa5995a912
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138281
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28093}
This gives assurance that we're not calling any function in
cricket::P2PTransportChannel off-thread.
Bug: none
Change-Id: I21d4e496cf5f301ab85abbd53a5abd4f5068ec39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138271
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28077}
- Implement datagram transport adaptor, which wraps datagram transport in DtlsTransportInternal. Datagram adaptor owns both ICE and Datagram Transports.
- Implement setup of datagram transport based on RTCConfiguration flag use_datagram_transport. This is very similar to MediaTransport setup with the exception that we create DTLS datagram adaptor.
- Propagate maximum datagram size to video encoder via MediaTransportConfig.
TODO: Currently this CL can only be tested in downstream projects. Once we add fake datagram transport, we will be able to implement unit tests similar to loopback media transport.
Bug: webrtc:9719
Change-Id: I4fa4a5725598dfee5da4f0f374269a7e289d48ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138100
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28047}
The gathering of host candidates with mDNS names is asynchronous and its
completion can happen after a srflx candidate is gathered by the same
underlying socket. We have a broken check in UDPPort::CreateConnection()
that assumes the gathering of host and srflx candidates is sequential.
This CL also does minor refactoring and clean-up.
Bug: chromium:944577
Change-Id: Ic28136a9515081f40b232a22fcbf4209814ed33a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138043
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28030}
This is a reland of e779847fb6499ac2dc4757de8c625ac377e9d0d4
Original change's description:
> Change SimpleStringBuilder::Append to not use strcpyn and SIZE_UNKNOWN
>
> Also add explicit includes of rtc_base/string_utils.h in files depending on it.
>
> Bug: webrtc:6424
> Change-Id: Id6b53937ab2d185d092a5d8863018fd5f1a88e27
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135744
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27903}
Tbr: kwiberg@webrtc.org
Bug: webrtc:6424
Change-Id: Ic08d5d7fbc25ff89e4182d7c9cb3b0e8e356339a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135946
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27957}
Also add explicit includes of rtc_base/string_utils.h in files depending on it.
Bug: webrtc:6424
Change-Id: Id6b53937ab2d185d092a5d8863018fd5f1a88e27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135744
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27903}
Methods of P2PTransportChannel have been assuming a non-null port
allocator for a long time, and yet the constructor does not check for
that. With the recent change that wires a signal in the port allocator
to the transport in the constructor, a valid allocator becomes a must.
Bug: None
Change-Id: I4ec2e5b577d74a598ee3c2f8ad59e9f0285ac4b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135880
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27897}
This is a reland of cd8d1cf68e4eeed71fba51c97006a91bfd41813d
Original change's description:
> Surface ICE candidates that match an updated candidate filter.
>
> After this change an ICE agent can surface candidates that do not match
> the previous filter but are allowed by the updated one. The candidate
> filter, as part of the internal implementation in the ICE transport,
> manifests the RTCIceTransportPolicy field in RTCConfiguration.
>
> This new feature would allow an ICE agent to gather new candidates when
> the transport policy changes from e.g. 'relay' to 'all' without an ICE
> restart.
>
> A caveat in the current implementation remains, and a candidate can
> surface multiple times if the transport policy, or the candidate filter
> directly, performs multiple transitions from a value that disallows to
> one that allows the underlying candidate type. For example, if the
> transport policy is updated by 'all' -> 'relay' -> 'all', the same host
> candidate can surface after the second update.
>
>
> Bug: webrtc:8939
> Change-Id: I92c2e07dafab225c702c5de28f47958a0d3270cc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132282
> Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27674}
Bug: webrtc:8939
Change-Id: I9c32b1ea05028ecd937ab4912779dd958faf734f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133582
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27694}
This reverts commit cd8d1cf68e4eeed71fba51c97006a91bfd41813d.
Reason for revert: breaks an internal project
Original change's description:
> Surface ICE candidates that match an updated candidate filter.
>
> After this change an ICE agent can surface candidates that do not match
> the previous filter but are allowed by the updated one. The candidate
> filter, as part of the internal implementation in the ICE transport,
> manifests the RTCIceTransportPolicy field in RTCConfiguration.
>
> This new feature would allow an ICE agent to gather new candidates when
> the transport policy changes from e.g. 'relay' to 'all' without an ICE
> restart.
>
> A caveat in the current implementation remains, and a candidate can
> surface multiple times if the transport policy, or the candidate filter
> directly, performs multiple transitions from a value that disallows to
> one that allows the underlying candidate type. For example, if the
> transport policy is updated by 'all' -> 'relay' -> 'all', the same host
> candidate can surface after the second update.
>
>
> Bug: webrtc:8939
> Change-Id: I92c2e07dafab225c702c5de28f47958a0d3270cc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132282
> Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27674}
TBR=shampson@webrtc.org,qingsi@webrtc.org,jeroendb@webrtc.org,sukhanov@webrtc.org
Change-Id: Idd51a640e55a612b42fe8b69e05dff57a22d021a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133581
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27677}
After this change an ICE agent can surface candidates that do not match
the previous filter but are allowed by the updated one. The candidate
filter, as part of the internal implementation in the ICE transport,
manifests the RTCIceTransportPolicy field in RTCConfiguration.
This new feature would allow an ICE agent to gather new candidates when
the transport policy changes from e.g. 'relay' to 'all' without an ICE
restart.
A caveat in the current implementation remains, and a candidate can
surface multiple times if the transport policy, or the candidate filter
directly, performs multiple transitions from a value that disallows to
one that allows the underlying candidate type. For example, if the
transport policy is updated by 'all' -> 'relay' -> 'all', the same host
candidate can surface after the second update.
Bug: webrtc:8939
Change-Id: I92c2e07dafab225c702c5de28f47958a0d3270cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132282
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27674}
This is used to avoid thread processing in simulated time
controller. This saves up to 30% execution time in debug builds.
Bug: webrtc:10365
Change-Id: Ie83dfb2468d371e4687d28c776acf7e23eb411d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133173
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27666}
This patch introduces a minor tweak to how often
the extra ice pings are sent.
- never send if non of the candidates is relay
- only send (extra) if it was more than 100ms
since you sent a ping.
The motivation for this is that we measured
an regression of 0.05% in call setup success rate.
Bug: webrtc:10273
Change-Id: Icff36297d57030853a9ff8d4f74aaf6c84051d26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132702
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27601}
This allows the ciphersuite to be accessed from Blink code, which is
useful if we want to emit deprecation messages.
Bug: none
Change-Id: Idcf1f5402948406e4cf7c6e54b67a622a1a403ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132004
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27581}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27410}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
Some NaCl system headers live in a special directory and the
toolchain doesn't propagate the -I compiler flag [2].
A common workaround in Chromium is to use 'public_deps' in order
to propagate //native_client_sdk/src/libraries/nacl_io:nacl_io_include_dirs
one step further in the build graph.
[1] - https://cs.chromium.org/chromium/src/native_client_sdk/src/libraries/nacl_io/
[2] - -Inative_client_sdk/src/libraries/third_party/newlib-extras
Bug: chromium:925028
Change-Id: I5145b80c2ae6969f79fcbfcf93a6b05c8a122746
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27373}
This patch adds a field trial for appending an extra
attribute which contains the remote ufrag for the permission.
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
Bug: webrtc:10350
Change-Id: I7d62ab94e947bddd670d8eb53d4ff89b1d1e275c
Reviewed-on: https://webrtc-review.googlesource.com/c/123901
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26859}
Currently the IceTransportState goes to connected before the connection
is writable and also goes to checking before any candidate pairs are
being checked. This change fixes these cases.
Bug: webrtc:10352, chromium:933802
Change-Id: I64a67c7f76a94a6be9da413740ddc8762fe966ce
Reviewed-on: https://webrtc-review.googlesource.com/c/124102
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26842}
As a unintended consequence of the changed iceConnectionState implementation we won't ever transition to the "failed" iceConnection state unless a connection has first been established. This CL fixes that and adds a integration test for this scenario.
Bug: chromium:933786
Change-Id: I45effd7411959ac0e5b16a13d7568756dbeff4d1
Reviewed-on: https://webrtc-review.googlesource.com/c/123785
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26811}
