instead use the standard API to get the rollover counter and
determine the extended sequence number which is the basis for the packet index.
See https://github.com/cisco/libsrtp/issues/738 and
https://github.com/cisco/libsrtp/issues/721
BUG=webrtc:357776213
Change-Id: I90c5a4a538f56132158aa48db8700187fcdb47d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371960
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43802}
This is a reland of commit 9572b2fa5850da6d319b9efb5ee36290e2895f7f
that does not remove the legacy implementations yet.
Original change's description:
> srtp: spanify Protect + Unprotect
>
> Makes SrtpSession and SrtpTransport use rtc::CopyOnWriteBuffer for the Protect and Unprotect operations instead of passing around void pointers.
>
> Also updates the unit tests to use CopyOnWriteBuffer instead of char arrays with a fixed length.
>
> BUG=webrtc:357776213
> No-Iwyu: missing include is a private libsrtp header
>
> Change-Id: I02a22ceb4e183e93c4ebd8c0a9c931404e0e32f3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358442
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Cr-Commit-Position: refs/heads/main@{#43601}
No-Iwyu: missing include is a private libsrtp header
Bug: webrtc:357776213
Change-Id: I93704e27a6c48e015b775712fcd848c8c0c753e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372321
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43799}
This reverts commit 9572b2fa5850da6d319b9efb5ee36290e2895f7f.
Reason for revert: Breaks internal tests
Original change's description:
> srtp: spanify Protect + Unprotect
>
> Makes SrtpSession and SrtpTransport use rtc::CopyOnWriteBuffer for the Protect and Unprotect operations instead of passing around void pointers.
>
> Also updates the unit tests to use CopyOnWriteBuffer instead of char arrays with a fixed length.
>
> BUG=webrtc:357776213
> No-Iwyu: missing include is a private libsrtp header
>
> Change-Id: I02a22ceb4e183e93c4ebd8c0a9c931404e0e32f3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358442
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Cr-Commit-Position: refs/heads/main@{#43601}
Bug: webrtc:357776213
Change-Id: I5c36ecc2fd9ab672f61cd6b15398452cbd5e98a8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372200
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43608}
Makes SrtpSession and SrtpTransport use rtc::CopyOnWriteBuffer for the Protect and Unprotect operations instead of passing around void pointers.
Also updates the unit tests to use CopyOnWriteBuffer instead of char arrays with a fixed length.
BUG=webrtc:357776213
No-Iwyu: missing include is a private libsrtp header
Change-Id: I02a22ceb4e183e93c4ebd8c0a9c931404e0e32f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358442
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43601}
https://webrtc.googlesource.com/src/+/7738bc23ed7fee0d4856bdfe7b88985865829441
switched from using sizeof(uint32_t) to SRTP_SRCTP_INDEX_LEN.
It turned out that this is not always defined.
This patch defines it to 4.
BUG=webrtc:42222036
Change-Id: Ice3d24a6300d19bc2f573469aadd6474ace1b147
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43548}
This reverts commit 7738bc23ed7fee0d4856bdfe7b88985865829441.
Reason for revert: Some downstream projects are still using an older version of libsrtp
Original change's description:
> srtp: use SRTP_SRCTP_INDEX_LEN define from libsrtp 2.6.0
>
> BUG=webrtc:42222036
>
> Change-Id: Ibf5c6b200501c114b9709b76685bb0ecd30bf9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359627
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43538}
Bug: webrtc:42222036
Change-Id: Icdac768bd4ccb6f1f4ada68637c0b979aefc39f6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371240
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43544}
which may show useful debug logging.
Also document that we need to forward-declare the internal srtp_ctx_
struct instead of srtp_t.
BUG=webrtc:361372443
Change-Id: I76b1a4fb385af0fc1532f0ce6d0692b804f003dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360182
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43022}
This uses libSRTPs srtp_remove_stream()
https://github.com/cisco/libsrtp/blob/main/include/srtp.h#L597
method to remove SSRCs from the libSRTP session when they are removed
from the RTP demuxer. This works even when the stream was added
automatically via the ssrc_any_inbound mechanism.
Only streams for inbound SSRCs that were added explicitly via SDP negotiation are removed.
Guarded by WebRTC-SrtpRemoveReceiveStream field trial.
BUG=webrtc:15604
Change-Id: I655bde5f8ddf26ac91395ef54bd1b3c598813380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324720
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41105}
Avoids using webrtc::GlobalMutex. Since state is allocated on first
use and never destroyed, we avoid an exit-time destructor when
building with absl::Mutex.
Bug: webrtc:11567
Change-Id: Ib9c6480ab0474e37a853460115b35d961b93009c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258080
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36455}
This cl/
1) move WebRtcKeyValueConfig from api/transport to api/ directory.
2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials).
3) removes usage of webrtc::field_trial:: from the pc/ directory.
4) removes a few unused includes of system_wrappers/field_trial.h.
Bug: webrtc:10335
Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36160}
Also apply IWYU to all .cc files in pc/, and correct BUILD file to match.
Note: Some files came out wrong when iwyu was applied. These are not included.
Bug: none
Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36064}
rtp_rtcp_format is lighter build target than rtc_media_base and
a more natural place to keep rtp parsing functions.
Bug: None
Change-Id: Ibcb5661cc65edbdc89a63f3e411d7ad1218353cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226330
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34504}
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}
use the helper functions
srtp_crypto_policy_set_from_profile_for_rtp
and
srtp_crypto_policy_set_from_profile_for_rtcp
provided by libsrtp to initialize the rtp and rtcp policies.
BUG=None
Change-Id: Ib1560c0fc1c06d9e79c1f871b028555b3b4d66d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33399}
documents why it is safe to not follow libsrtp's advice
to ensure additional SRTP_MAX_TRAILER_LEN bytes are available
when calling srtp_protect (and similar srtcp functions).
BUG=None
Change-Id: I504645d21553160f06133fd8bb3ee79e178247da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209064
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33396}
which was missing a setfill call, leading to invalid timestamps.
BUG=webrtc:10675
Change-Id: Ib60f9f18b250aa89103e8de70b525df13c1042bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205780
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33183}
guarded by a new field trial flag WebRTC-Debugging-RtpDump.
Packets have a RTP_DUMP postfix for easy grep-ing.
BUG=webrtc:10675
Change-Id: I73c0e0db47dca1079cd303c41a8b80fd7ae4a902
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196087
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32775}
This change migrates a last stray consumer of GlobalLock
(SrtpSession) and removes all traces of GlobalLock/GlobalLockScope
from WebRTC.
Bug: webrtc:11567
Change-Id: I28059f2a10075815a4bdee8c357b9d3b6e50f18b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179361
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31736}
Merge GlobalLock and GlobalLockPod, make member private.
annotate creation of all GlobalLocks with ABSL_CONST_INIT
Bug: None
Change-Id: I29abcc86796ec0e45b15df7d26392309d1bf7324
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156303
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29447}
