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WWebRTC LUCI CQdcsctp: Exit deferred stream reset on FORWARD-TSN
06fbe63c创建于 2023年9月26日历史提交
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BUILD.gn
sctp: Fix data channel closing sequence When an SCTP stream is closing, a stream reset needs to be sent from both ends. The remote was not sending a stream reset and quickly opening another stream with the same StreamID could cause SCTP errors. Bug: webrtc:13994 Change-Id: I3abc74ddc88b3fcf7e6495d76e7d77f52280b5d1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260922 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36773} 3 年前
dcsctp_handover_state.cc
dcsctp: move HandoverReadinessStatus::ToString definition to build target public:socket This allows build targets that need only HandoverReadinessStatus to depend only on public:socket, and not on socket:dcsctp_socket. Bug: webrtc:13154 Change-Id: I29f41910cdb5baed96b57fd7284b96fc50a56ba4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232331 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35037} 4 年前
dcsctp_handover_state.h
dcsctp: Add handover state for zero checksum This CL can prepare downstream projects for being aware of this new handover state. This was extracted from change 299076. Bug: webrtc:14997 Change-Id: I35bfbe040ffbaa5d7266eb67d58078b66083337a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302980 Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39927} 3 年前
dcsctp_message.h
Revert "dcsctp: Use rtc::CopyOnWriteBuffer" This reverts commit 2db59a6584eca54245794a0e657ca9ded9e6707f. Reason for revert: Causes msan-issue in crc32c, reading uninitialized memory. Bug: webrtc:12943, chromium:1275559 Change-Id: I05f1012d896aeaca86c4562e0df15fa7ea326d60 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239560 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35461} 4 年前
dcsctp_options.h
Reland "dcsctp: Support zero checksum packets" This reverts commit 45eae346930aedbf4624d054e0633c94b397b8ec. It was found not to be the root cause of the performance regression, so it's safe to reland. Bug: webrtc:14997 Change-Id: I67c90752875bf4071cbdd5adfa462a37f4d4ceab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302162 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#39910} 3 年前
dcsctp_socket.h
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} 2 年前
dcsctp_socket_factory.cc
sctp: Fix data channel closing sequence When an SCTP stream is closing, a stream reset needs to be sent from both ends. The remote was not sending a stream reset and quickly opening another stream with the same StreamID could cause SCTP errors. Bug: webrtc:13994 Change-Id: I3abc74ddc88b3fcf7e6495d76e7d77f52280b5d1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260922 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36773} 3 年前
dcsctp_socket_factory.h
sctp: Fix data channel closing sequence When an SCTP stream is closing, a stream reset needs to be sent from both ends. The remote was not sending a stream reset and quickly opening another stream with the same StreamID could cause SCTP errors. Bug: webrtc:13994 Change-Id: I3abc74ddc88b3fcf7e6495d76e7d77f52280b5d1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260922 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36773} 3 年前
mock_dcsctp_socket.h
dcsctp: Add public API for setting priorities This is a reland of commit 17a02a31d7d2897b75ad69fdac5d10e7475a5865. This is the first part of supporting stream priorities, and adds the API and very basic support for setting and retrieving the stream priority. This commit doesn't in any way change the actual packet sending - the specified priority values are stored, but not acted on. This is all that is client visible, so clients can start using the API as written, and they would never notice that things are missing. Bug: webrtc:5696 Change-Id: I04d64a63cbaec67568496ad99667e14eba85f2e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264424 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37081} 3 年前
mock_dcsctp_socket_factory.h
sctp: Fix data channel closing sequence When an SCTP stream is closing, a stream reset needs to be sent from both ends. The remote was not sending a stream reset and quickly opening another stream with the same StreamID could cause SCTP errors. Bug: webrtc:13994 Change-Id: I3abc74ddc88b3fcf7e6495d76e7d77f52280b5d1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260922 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36773} 3 年前
mock_dcsctp_socket_test.cc
dcsctp: Add mocks This is for convenience to the users of dcSCTP, which may want to have unit tests where the socket is mocked. And since it's best practice not to mock other teams' or project's classes, a mock will be provided by the upstream project - this one. Bug: webrtc:12614 Change-Id: I65d5d21097e7feda9162567560d3838759c962fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224161 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34385} 4 年前
packet_observer.h
dcsctp: Add public API Clients will use this API for all their interactions with this library. It's made into an interface (of which there will only be a single implementation) simply for documentation purposes. But that also allows clients to mock the library if wanted or to have a thread-safe wrapper, as the library itself is not thread-safe, while keeping the same interface. Bug: webrtc:12614 Change-Id: I346af9916e26487da040c01825c2547aa7a5d0b7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213348 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33648} 5 年前
text_pcap_packet_observer.cc
dcsctp: Extract logging packet observer as utility It is useful for more than just the transport. Bug: webrtc:12961 Change-Id: Iad064c8fb707ca589a1c232e17436338fb06623d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225543 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34451} 4 年前
text_pcap_packet_observer.h
dcsctp: Extract logging packet observer as utility It is useful for more than just the transport. Bug: webrtc:12961 Change-Id: Iad064c8fb707ca589a1c232e17436338fb06623d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225543 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34451} 4 年前
timeout.h
dcsctp: Add public API Clients will use this API for all their interactions with this library. It's made into an interface (of which there will only be a single implementation) simply for documentation purposes. But that also allows clients to mock the library if wanted or to have a thread-safe wrapper, as the library itself is not thread-safe, while keeping the same interface. Bug: webrtc:12614 Change-Id: I346af9916e26487da040c01825c2547aa7a5d0b7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213348 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33648} 5 年前
types.h
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} 2 年前
types_test.cc
dcsctp: Add operators on TimeMs and DurationMs To be able to use them type-safely, they should support native operators (e.g. adding a time and a duration, or subtracting two time values), as the alternative is to manage them as numbers. Yes, this makes them behave a bit like absl::Time/absl::Duration. Bug: webrtc:12614 Change-Id: I4dea12e33698a46e71fb549f44c06f2f381c9201 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215143 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33725} 5 年前