Patch originally submitted by Lennart Grahl:
https://webrtc-review.googlesource.com/c/src/+/177527
SCTP message chunks and notifications are being delivered interleaved.
However, the way the code was structured previously, a notification
would interrupt reassembly of a message chunk and hand out the partial
message, thereby violating message integrity. This patch separates the
handling of notifications and reassembly of messages.
Additional changes:
- Remove illegal cast from non-validated u32 to enum (PPID)
- Drop partial messages if the SID has been changed but EOR not yet
received instead of delivering them. (This should never happen
anyway.)
- Don't treat TSN as timestamp (wat)
- Replace "usrsctplib/usrsctp.h" with <usrsctp.h>, allowing a hack
to be removed from media/BUILD.gn
Bug: webrtc:11708
Change-Id: I29733b03f67a3d840104b8608a7f0083466c2d0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178469
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31641}
This reverts commit 6cbff752f52bf3f70168d551c33ce719bd8e0663.
Reason for revert: breaking downstream projects, Win MSVC x86 dbg and Win x86 Clang rel
Original change's description:
> Fix data channel message integrity violation
>
> SCTP message chunks and notifications are being delivered interleaved.
> However, the way the code was structured previously, a notification
> would interrupt reassembly of a message chunk and hand out the partial
> message, thereby violating message integrity. This patch separates the
> handling of notifications and reassembly of messages.
>
> Additional changes:
>
> - Remove illegal cast from non-validated u32 to enum (PPID)
> - Drop partial messages if the SID has been changed but EOR not yet
> received instead of delivering them. (This should never happen
> anyway.)
> - Don't treat TSN as timestamp (wat)
>
> Bug: webrtc:11708
> Change-Id: I4e2fe2262feda2a96d2ae3f6ce9b06370d9878ae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177527
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31605}
TBR=deadbeef@webrtc.org,kwiberg@webrtc.org,tommi@webrtc.org,hta@webrtc.org,lennart.grahl@gmail.com
Change-Id: I6d6c5a11835f155f8c449b996d034f43b8db452c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11708
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178488
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31606}
SCTP message chunks and notifications are being delivered interleaved.
However, the way the code was structured previously, a notification
would interrupt reassembly of a message chunk and hand out the partial
message, thereby violating message integrity. This patch separates the
handling of notifications and reassembly of messages.
Additional changes:
- Remove illegal cast from non-validated u32 to enum (PPID)
- Drop partial messages if the SID has been changed but EOR not yet
received instead of delivering them. (This should never happen
anyway.)
- Don't treat TSN as timestamp (wat)
Bug: webrtc:11708
Change-Id: I4e2fe2262feda2a96d2ae3f6ce9b06370d9878ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177527
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31605}
Tests the behavior of the usrsctp library buffering a large message in
unordered mode. The expected behavior is that this message will be sent
when the socket becomes unblocked, but instead an SCTP_SEND_FAILED_EVENT
is fired by usrsctp library and the message is never sent. This test
will pass with a newer version of usrsctp lib, or if the send is in
ordered mode.
Bug: webrtc:10939
Change-Id: I3b4b05e7dcc7574bf3397991848a9ad7122adc0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172480
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30950}
This is part of a CL series merging rtc::MessageQueue into rtc::Thread.
Bug: webrtc:9883
Change-Id: I3cb857cc707d5e897759366d1478cc1ec19bce9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165344
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30180}
Currently we set the EOR bit when sending a message through the sctp
library. This makes the send non atomic, meaning that message can be
partially accepted by the sctp socket. Currently we ignore the sent
amount result, but this change now checks that result and buffers the
remaining message to be sent later in the case that it was only
partially accepted by usrsctp.
Bug: webrtc:10922
Change-Id: I9ff563c40e2b7dbdeb19b40d07c43a15ff7c9b49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150562
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29051}
This also changes the default when no max-message-size is set
to the protocol defined value of 64K, and prevents messages
from being sent when they are too large to send.
Bug: webrtc:10358
Change-Id: Iacc1dd774d1554d9f27315378fbea6351300b5cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135948
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27945}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
