Upstream commit: https://webrtc.googlesource.com/src/+/a93f581705787105ac28bccdc179778fd4400298
dcsctp: Don't generate FORWARD-TSN across stream resets
This was a fun bug which proved to be challenging to find a good
solution for. The issue comes from the combination of partial
reliability and stream resetting, which are covered in different RFCs,
and where they don't refer to each other...
Stream resetting (RFC 6525) is used in WebRTC for closing a Data
Channel, and is done by signaling to the receiver that the stream
sequence number (SSN) should be set to zero (0) at some time. Partial
reliability (RFC 3758) - and expiring messages that will not be
retransmitted - is done by signaling that the SSN should be set to a
certain value at a certain TSN, as the messages up until the provided
SSN are not to be expected to be sent again.
As these two functionalities both work by signaling to the receiver
what the next expected SSN should be, they need to do it correctly not
to overwrite each others' intent. And here was the bug. An example
scenario where this caused issues, where we are Z (the receiver),
getting packets from the sender (A):
5 A->Z DATA (TSN=30, B, SID=2, SSN=0)
6 Z->A SACK (Ack=30)
7 A->Z DATA (TSN=31, E, SID=2, SSN=0)
8 A->Z RE_CONFIG (REQ=30, TSN=31, SID=2)
9 Z->A RE_CONFIG (RESP=30, Performed)
10 Z->A SACK (Ack=31)
11 A->Z DATA (TSN=32, SID=1)
12 A->Z FORWARD_TSN (TSN=32, SID=2, SSN=0)
Let's assume that the path Z->A had packet loss and A never really
received our responses (#6, #9, #10) in time.
At #5, Z receives a DATA fragment, which it acks, and at #7 the end of
that message. The stream is then reset (#8) which it signals that it
was performed (#9) and acked (#10), and data on another stream (2) was
received (#11). Since A hasn't received any ACKS yet, and those chunks
on SID=2 all expired, A sends a FORWARD-TSN saying that "Skip to TSN=32,
and don't expect SID=2, SSN=0". That makes the receiver expect the SSN
on SID=2 to be SSN=1 next time at TSN=32.
But that's not good at all - A reset the stream at #8 and will want to
send the next message on SID=2 using SSN=0 - not 1. The FORWARD-TSN
clearly can't have a TSN that is beyond the stream reset TSN for that
stream.
This is just one example - combining stream resetting and partial
reliability, together with a lossy network, and different variants of
this can occur, which results in the receiver possibly not delivering
packets because it expects a different SSN than the one the sender is
later using.
So this CL adds "breakpoints" to how far a FORWARD-TSN can stretch. It
will simply not cross any Stream Reset last assigned TSNs, and only when
a receiver has acked that all TSNs up till the Stream Reset last
assigned TSN has been received, it will proceed expiring chunks after
that.
Bug: webrtc:14600
Change-Id: Ibae8c9308f5dfe8d734377d42cce653e69e95731
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321600
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40829}
Essentially a no-op since we're going to see this change
reverted when we vendor in c2bbe4b952.
Upstream commit: https://webrtc.googlesource.com/src/+/d8633868b34dc1d841f0a9fd1afe2bc22aa8bde6
Enable SRTP GCM ciphers by default
Bug: webrtc:15178
Change-Id: I0216ce8f194fffc820723d82b9c04a76573c2f4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305381
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40828}
Upstream commit: https://webrtc.googlesource.com/src/+/98e71f57eaa94dc085e712d939c19c0bf57b9b37
Subtract an additional 5kbps of the bitrate when backing off.
Traditionally, we'd back off to 85% of the measured throughput in response to an overuse. However, this backoff doesn't appear to be sufficient to drain the queues in some low-bitrate scenarios, and the problem has gotten a bit worse with the RobustThroughputEstimator. (The new estimate looks more stable. The old estimator had more variation, the lowest points were lower, causing backoffs to lower rates.)
With this change, we back off to 0.85*thoughput-5kbps. The difference is negligible except at low bitrates.
Bug: webrtc:13402,b/298636540
Change-Id: I53328953c056b8ad77f6c7561d6017f171b2dfbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321701
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40827}
Upstream commit: https://webrtc.googlesource.com/src/+/f97058e9edffe109ecf1cec94173916691f42809
Move static functions in media_session into anonymous namespace
and clean up methods that are now detected as unused.
BUG=None
Change-Id: If5dac3d43d4df6c7c108504c202c2383fe4a3f27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321580
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40823}
Upstream commit: https://webrtc.googlesource.com/src/+/d12e75997cfb814e8e4fa6a07d31f797643d53eb
Add instructions for adding and removing field trials
Since the registry now also stores the end date, the bug doesn't have to
specify this.
Bug: webrtc:14154
Change-Id: I94dee43105079607ff9d820e238018304eb441a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321582
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40821}
We already cherry-picked this when we vendored d8f2b0380b.
Upstream commit: https://webrtc.googlesource.com/src/+/83894d384763c613e548e6352838406e6e0c2fc1
Fire MaybeSignalReadyToSend in a PostTask when recursive
Speculative fix. Writing the test for it is more complex.
Bug: chromium:1483874
Change-Id: Icfaf1524b0499c609010753e0b6f3cadbd0e98f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321480
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40820}
Upstream commit: https://webrtc.googlesource.com/src/+/4001cc74533088b6a10ff32614df3b1264b46cb8
Populate field trial registry
The lists has been constructed by grepping the code base from commit
bfc2a3553d56 ("Remove more codec-related templating") using the PCRE
'(?<=")WebRTC-[^\s/"]+' and manually removing some false positives. Each
field trials has then, on a best effort basis, been associated with a
bug by doing a reverse git log lookup and using the corresponding bug
tag that was associated with the commit where the field trial key was
first introduced.
The field trials are divided into active and policy exempt. The latter
are for field trials that were added before commit c4a35898d916 ("Add
documentation for field trials") which introduced the new policy that
field trials henceforth needs to be registered. These field trials may
not have an associated bug nor an end date.
For all field trials that have been deemed experimentational, including
many policy exempt ones, an initial end date of 2024-04-01 has been
chosen. This date was chosen based on that the policy was introduced on
2022-06-23 and will give about 6 month grace period to allow cleanup of
potentially already expired field trials. Owners are of course free to
adjust the end date at a later time.
The lists have been validated by running most tests with the following
GN arg set:
rtc_strict_field_trials = "dcheck"
Bug: webrtc:14154
Change-Id: Ic86d96933fbe0f18393ec57d36719a382855c694
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321581
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40819}
Upstream commit: https://webrtc.googlesource.com/src/+/0505115b5c80162d84b1c024c31515457a4438fc
Pass the correct abs_capture_timestamp while cloning audio frame
This change replaces type of absolute_capture_timestamp_ms_ in
TransformableOutgoingAudioFrame from int to optional uint and makes
the function AbsoluteCaptureTimestamp() inside
TransformableAudioFrameInterface pure virtual.
Bug: webrtc:14949
Change-Id: Id3bdbcba63a5f91105ab198208e4f2b11eb3c7db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319000
Commit-Queue: Palak Agarwal <agpalak@google.com>
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40814}
Upstream commit: https://webrtc.googlesource.com/src/+/06fbe63cbf81c7731e1fcd68f26aa064953f3347
dcsctp: Exit deferred stream reset on FORWARD-TSN
https://datatracker.ietf.org/doc/html/rfc6525#section-5.2.2:
E2: If the Sender's Last Assigned TSN is greater than the cumulative
acknowledgment point, then the endpoint MUST enter "deferred
reset processing". ... until the cumulative
acknowledgment point reaches the Sender's Last Assigned TSN.
The cumulative acknowledgement point can not only be reached by
receiving DATA chunks, but also by receiving a FORWARD-TSN that
instructs the receiver to skip them. This was only done for DATA and not
for FORWARD-TSN, which is now corrected.
Additionally, an unnecessary implicit sending of SACK after having
received FORWARD-TSN was removed as this is done anyway every time a
packet has been received. This unifies the processing of DATA and
FORWARD-TSN more.
Bug: webrtc:14600
Change-Id: If797d3c46e741074fe05e322d0aebec765a87968
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40811}
We already cherry-picked this when we vendored 2f4bc64166.
Upstream commit: https://webrtc.googlesource.com/src/+/b4d4bbcebdfef35a8623268656001558beae1382
Revert "Clean up last_packet_received_time_ as it's no longer used."
This reverts commit 2f4bc6416651be40ef8f95a4695e6b7c41f18666.
Reason for revert: Breaks downstream test
Original change's description:
> Clean up last_packet_received_time_ as it's no longer used.
>
> Bug: webrtc:15377
> Change-Id: I5453b9fd572a04dbea3241a2eb1c8ad8bb8b1186
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320560
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Ying Wang <yinwa@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40792}
Bug: webrtc:15377
Change-Id: Ifa57671cc479cdd86f543c4edc236221beb76f90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321340
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40797}
Upstream commit: https://webrtc.googlesource.com/src/+/9c58483b5aaed6e83cedc953ae0ae1934e7dcedb
Rename EncodedImage property Timetamp to RtpTimestamp
To avoid name collision with Timestamp type,
To avoid confusion with capture time represented as Timestamp
Bug: webrtc:9378
Change-Id: I8438a9cf4316e5f81d98c2af9dc9454c21c78e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40796}
Upstream commit: https://webrtc.googlesource.com/src/+/850296b7a4428f2a99803daaedeaea2016fb1671
Reapply "dcsctp: Negotiate zero checksum"
The handover state has been added with
commit daaa6ab5a8c74b87d9d6ded07342d8a2c50c73f7.
This reverts commit 014cbed9d2377ec0a0b15f2c48b21a562f770366.
Bug: webrtc:14997
Change-Id: Ie84f3184f3ea67aaa6438481634046ba18b497a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320941
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40794}
Essentially a no-op since we're going to see this change
reverted when we vendor in b4d4bbcebd.
Upstream commit: https://webrtc.googlesource.com/src/+/2f4bc6416651be40ef8f95a4695e6b7c41f18666
Clean up last_packet_received_time_ as it's no longer used.
Bug: webrtc:15377
Change-Id: I5453b9fd572a04dbea3241a2eb1c8ad8bb8b1186
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320560
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40792}
We already cherry-picked this when we vendored b6c7ddd6a1.
Upstream commit: https://webrtc.googlesource.com/src/+/4aa2b40ffeab51e3b95561bf2ff53137c662105c
Revert "Use loss based bwe v2 in the start phase."
This reverts commit b6c7ddd6a137e187fa459255488da3b70b0a6c24.
Reason for revert: broken unit test
Original change's description:
> Use loss based bwe v2 in the start phase.
>
> TESTED=manual before:screen/ANtkMApoYczA2V5; after:screen/9kBoSvYKzKZR4sK
>
> Bug: webrtc:12707
> Change-Id: Ic156e363625c4b7476011059f3cd95641972091c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320840
> Commit-Queue: Diep Bui <diepbp@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40789}
Bug: webrtc:12707
Change-Id: Ibde45436934707b8e0084aa496dc249bc1c78ab2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321180
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#40790}
Essentially a no-op since we're going to see this change
reverted when we vendor in 4aa2b40ffe.
Upstream commit: https://webrtc.googlesource.com/src/+/b6c7ddd6a137e187fa459255488da3b70b0a6c24
Use loss based bwe v2 in the start phase.
TESTED=manual before:screen/ANtkMApoYczA2V5; after:screen/9kBoSvYKzKZR4sK
Bug: webrtc:12707
Change-Id: Ic156e363625c4b7476011059f3cd95641972091c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320840
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40789}
