0
代码
代码
Issues
Pull Requests
讨论
Wiki
分析
0
WWebRTC LUCI CQdcsctp: Abandon correct message on stream reset
9cf825de创建于 2023年10月6日历史提交
文件最后提交记录最后更新时间
BUILD.gn
[Unwrap] Migrate dcsctp sequence numbers to SeqNumUnwrapper Bug: webrtc:13982 Change-Id: Ic900a967d1b8e96a2b1ec99424674ccb33eb7165 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288940 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39084} 3 年前
handover_testing.cc
dcsctp: support handover state serialization testing dcSCTP library users can set their custom g_handover_state_transformer_for_test that can serialize and deserialize the state. All dcSCTP handover tests call g_handover_state_transformer_for_test. If some part of the state is serialized incorrectly or is forgotten, high chance that it will fail the tests. Bug: webrtc:13154 Change-Id: I251a099be04dda7611e9df868d36e3a76dc7d1e1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232325 Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35035} 4 年前
handover_testing.h
dcsctp: support handover state serialization testing dcSCTP library users can set their custom g_handover_state_transformer_for_test that can serialize and deserialize the state. All dcSCTP handover tests call g_handover_state_transformer_for_test. If some part of the state is serialized incorrectly or is forgotten, high chance that it will fail the tests. Bug: webrtc:13154 Change-Id: I251a099be04dda7611e9df868d36e3a76dc7d1e1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232325 Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35035} 4 年前
internal_types.h
dcsctp: Abandon correct message on stream reset Before this CL, a message was identified by the triple (stream_id, is_unordered, MID) (and yes, the MID is always present in the send queue, even when interleaved message is not enabled.). So when a chunk was abandoned due to e.g. having reached the retransmission limit, all other chunks for that message in the retransmission queue, and all unsent chunks in the send queue were discarded as well. This works well, except for the fact that resetting a stream will result in the MID being set to zero again, which can result in two different messages having the same identifying triple. And due to the implementation, both messages would get abandoned. In WebRTC, an entire data channels is either reliable or unreliable, and for a message to be abandoned, the channel must be unreliable. So this means that in the case of stream resets - meaning that a channel was closed and then reopened, an abandoned message from the old (now closed) channel would result in abandoning another message sent on the re-opened data channel. This CL introduces a new internal property on messages while in the retransmission and send queue; The "outgoing message id". It's a monotonically increasing identifier - shared among all streams - that is never reset to zero in the event of a stream reset. And now a message is actually only identified by the outgoing message id, but often used together with the stream identifier, as all data in the send queue is partitioned by stream. This identifier is 32 bits wide, allowing at most four billion messages to be in-flight, which is not a limitation, as the TSN is also 32 bits wide. Bug: webrtc:14600 Change-Id: I33c23fb0e4bde95327b15d1999e76aa43f5fa7db Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322603 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40881} 2 年前
math.h
dcsctp: Ensure packet size doesn't exceed MTU Due to a previous refactoring, the SCTP packet header is only added when the first chunk is written. This wasn't reflected in the bytes_remaining, which made it add more than could fit within the MTU. Additionally, the maximum packet size must be even divisible by four as padding will be added to chunks that are not even divisble by four (up to three bytes of padding). So compensate for that. Bug: webrtc:12614 Change-Id: I6b57dfbf88d1fcfcbf443038915dd180e796191a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215145 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@{#33760} 5 年前
math_test.cc
dcsctp: Ensure packet size doesn't exceed MTU Due to a previous refactoring, the SCTP packet header is only added when the first chunk is written. This wasn't reflected in the bytes_remaining, which made it add more than could fit within the MTU. Additionally, the maximum packet size must be even divisible by four as padding will be added to chunks that are not even divisble by four (up to three bytes of padding). So compensate for that. Bug: webrtc:12614 Change-Id: I6b57dfbf88d1fcfcbf443038915dd180e796191a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215145 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@{#33760} 5 年前
sequence_numbers.h
[Unwrap] Migrate dcsctp sequence numbers to SeqNumUnwrapper Bug: webrtc:13982 Change-Id: Ic900a967d1b8e96a2b1ec99424674ccb33eb7165 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288940 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39084} 3 年前
sequence_numbers_test.cc
Move StrongAlias to rtc_base It's useful for other parts of WebRTC and there is no real reason why it should be located in net/dcsctp. Bug: None Change-Id: Iccaed4e943e21ddaea8603182d693114b2da9f6b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232606 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35055} 4 年前
str_join.h
dcsctp: Added common utilities These are quite generic utilities that are used by multiple modules within dcSCTP. Some would be good to have in rtc_base and are simple replicas of utilities available in abseil. Bug: webrtc:12614 Change-Id: I9914286ced7317a34628a71697da9149d6d19d38 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213190 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33609} 5 年前
str_join_test.cc
dcsctp: Added common utilities These are quite generic utilities that are used by multiple modules within dcSCTP. Some would be good to have in rtc_base and are simple replicas of utilities available in abseil. Bug: webrtc:12614 Change-Id: I9914286ced7317a34628a71697da9149d6d19d38 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213190 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33609} 5 年前