Many things are omitted in this doc and it can definitely be improved,
but I hope it captures the most important parts.
Bug: webrtc:12568
Change-Id: I13097d633ca19cecc9dd43bdb777b0ca48f151dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215142
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33724}
Previously, RTP header extensions with encryption had been filtered
if the encryption had been activated (not the other way around) which
was likely an unintended logic inversion.
In addition, it ensures that encrypted RTP header extensions are only
negotiated if RTP header extension encryption is turned on. Formerly,
which extensions had been negotiated depended on the order in which
they were inserted, regardless of whether or not header encryption was
actually enabled, leading to no extensions being sent on the wire.
Further changes:
- If RTP header encryption enabled, prefer encrypted extensions over
non-encrypted extensions
- Add most extensions to list of extensions supported for encryption
- Discard encrypted extensions in a session description in case encryption
is not supported for that extension
Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
header extensions will prevent any RTP packets being sent/received.
Bug: webrtc:11713
Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33723}
These are some fixes that were added after submission of
https://webrtc-review.googlesource.com/c/src/+/213664
Mainly:
* Don't accept TSNs that have a too large difference from expected
* Renaming of member variable (to confirm to style guidelines)
Bug: webrtc:12614
Change-Id: I06e11ab2acf5d307b68c3cbc135fde2c038ee690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215070
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33721}
Erle Uncertainty changes the residual echo computation during saturated
echo. However, the case of saturated echo is already handled by the
residual echo estimator causing the ErleUncertainty to be a no-op.
The change has been tested for bit-exactness.
Bug: webrtc:8671
Change-Id: I779ba67f99f29d4475a0465d05da03d42d50e075
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215072
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33719}
This also creates a g3doc directory under pc/
Bug: webrtc:12552
Change-Id: I0913c88831658776a0f02174b57b539ac85b4a9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215077
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33718}
As part of adding the new WgcCapturerWin implementation of the
DesktopCapturer interface, we should ensure that we can measure the
health and success of this new code. In order to quantify that, I've
added telemetry to measure the usage of each capturer implementation,
the time taken to capture a frame, and any errors that are encountered
in the new implementation.
I've also set the capturer id property of frames so that we can measure
error rates and performance of each implementation in Chromium as well.
This CL must be completed after this Chromium CL lands:
2806094: Add histograms to record new WebRTC DesktopCapturer telemetry | https://chromium-review.googlesource.com/c/chromium/src/+/2806094
Bug: webrtc:9273
Change-Id: I33b0a008568a4df4f95e705271badc3313872f17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214060
Commit-Queue: Austin Orion <auorion@microsoft.com>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33716}
* `AddTo` and `Difference` are made into static methods, as one may have
believed that these modified the current object previously. The
`Increment` method is kept, as it's obvious that it modifies the
current object as it doesn't have a return value, and `next_value` is
kept, as its naming (lower-case, snake) indicates that it's a simple
accessor.
* Difference will return the absolute difference. This is actually the
only reasonable choice, as the return value was unsigned and any
negative value would just wrap.
Bug: webrtc:12614
Change-Id: If14a71636e67fc612d12759dc80a9c2518c85281
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215069
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33714}
As of
1e4d4fdf88
we no longer expect an InitEncode on deativation of a layer.
Bug: webrtc:12540
Change-Id: I10d447d90d1019258f662caf7f6e649d63d6927a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215076
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33709}
Since we want most users to use the PeerConnection API, this is the
part that we should document.
If we want people to use other APIs, we can add to the file.
Bug: webrtc:12674
Change-Id: Icf14f218cf51c640e6f846f10b49dff84106dc21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215066
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33707}
Adding fuzzers to the build made "gn gen --check" discover a lot
of dependency errors between various components of dcSCTP.
Bug: webrtc:12614
Change-Id: I0b2dd7321aec2624da417f413c727bd11b4743e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215003
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33705}
This prepares for ability to defer sequence number assignment to after
the pacing stage - a scenario where the RtpRtcp module rather than than
RTPSender class has responsibility for sequence numbering.
Bug: webrtc:11340
Change-Id: Ife88f60258b9b7cfd9dbd3326f02ac34da8f7603
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214967
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33702}
No customers have been identified.
Bug: chromium:1197965
Change-Id: Ia3063d0909c718ffb8e824225c8c60180551115a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214963
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33700}
Following up on https://webrtc-review.googlesource.com/c/src/+/213000
This CL prevents scheduling work before TaskQueuePacedSender::EnsureStarted(),
making it necessary to function.
Bug: chromium:1152887
Change-Id: I848c9e6d6057a404626ad693b1f4dc7fba797a9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214320
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33695}
This causes build failures in the Chromium fuzzers, so let's disable it
for now.
Bug: none
Change-Id: I0a076c0cd5cfb7d62383d733f3934f8b58f8ad34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215040
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33693}
This is the result of compiling Chromium with
Wtautological-unsigned-zero-compare. For more details, see:
https://chromium-review.googlesource.com/c/chromium/src/+/2802412
Change-Id: I05cec6ae5738036a56beadeaa1dde5189edf0137
Bug: chromium:1195670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213783
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33689}
The rename ensures we don't confuse this implementation with
the new one based on the new dcSCTP library.
Bug: webrtc:12614
No-Presubmit: True
Change-Id: Ida08659bbea9c98aba8247d4368799ff7dd18729
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214482
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33688}
When migrating to use StrongAlias types, the PPID was incorrectly
modeled as an uint16_t instead of a uint32_t, as it was prior to using
StrongAlias. Most likely a copy-paste error from StreamID.
As the Data Channel PPIDs are in the range of 51-57, it was never caught
in tests.
Bug: webrtc:12614
Change-Id: I2b61ef7935df1222068e7f4e70fc2aaa532dcf7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214960
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33687}
simplifying the code and comparing against the value libsrtp expects
and increase verbosity of error logging related to key length mismatches.
BUG=None
Change-Id: Icc0d0121d2983e23c95b0f972a5f6cac1d158fd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213146
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33685}
This test flakes due to the expectation at
http://shortn/_XxN4cgzMLD.
Bug: webrtc:12590
Change-Id: Id75ecd4f12cd6f9af86aeb2213fd3cb39aecb6d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33684}
VideoStreamEncoderTest: Remove unneeded set_timestamp_rtp in CreateFrame methods (the timestamp is set based on ntp_time_ms in VideoStreamEncoder::OnFrame).
Bug: none
Change-Id: I6b5531a9ac21cde5dac54df6de9b9d43261e90c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214488
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33683}
The Reassembly Queue receives fragmented messages (DATA or I-DATA
chunks) and - with help of stream reassemblers - will reassemble these
fragments into messages, which will be delivered to the client.
It also handle partial reliability (FORWARD-TSN) and stream resetting.
To avoid a DoS attack vector, where a sender can send fragments in a way
that the reassembly queue will never succeed to reassemble a message and
use all available memory, the ReassemblyQueue has a maximum size.
Bug: webrtc:12614
Change-Id: Ibb084fecd240d4c414e096579244f8f5ee46914e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214043
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33678}
This class handles the assembly of fragmented received messages (as DATA
chunks) and manage per-stream queues. This class only handles
non-interleaved messages as described in RFC4960, and is not used when
message interleaving is enabled on the association, as described in
RFC8260.
This is also only part of the reassembly - a follow-up change will add
the ReassemblyQueue that handle the other part as well. And an even
further follow-up change will add a "interleaved reassembly stream".
Bug: webrtc:12614
Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33677}
The Data Tracker's purpose is to keep track of all received DATA chunks
and to ACK/NACK that data, by generating SACK chunks reflecting its view
of what has been received and what has been lost.
It also contains logic for _when_ to send the SACKs, as that's different
depending on e.g. packet loss. Generally, SACKs are sent every second
packet on a connection with no packet loss, and can also be sent on a
delayed timer.
In case partial reliability is used, and the transmitter has decided
that some data shouldn't be retransmitted, it will send a FORWARD-TSN
chunk, which this class also handles, by "forgetting" about those
chunks.
Bug: webrtc:12614
Change-Id: Ifafb0c211f6a47872e81830165ab5fc43ee7f366
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213664
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33676}
Each file is a SCTP packet (without any additional headers), all
extracted from a few Wireshark dumps that have been manually recorded.
Bug: webrtc:12614
Change-Id: I64bef0c563f1d83ae22735d702c8abafec6429b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214701
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33675}
