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11 commits

Author SHA1 Message Date
Harald Alvestrand
5cb7807a36 Implement crypto stats on DTLS transport 
Bug: chromium:1018077
Change-Id: I585d4064f39e5f9d268b408ebf6ae13a056c778a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158403
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29628}
 2019-10-28 11:30:23 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique. 
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
 2019-09-17 15:47:29 +00:00
Joachim Reiersen
637bed5f8d Add missing BoringSSL ifdef to OpenSSLStreamAdapter 
Compiling without BoringSSL fails since g_use_time_callback_for_testing
is defined inside a OPENSSL_IS_BORINGSSL block.

Bug: webrtc:10160
Change-Id: I25c27fa8ed128a50aa855db2012026c97954b91b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134226
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27779}
 2019-04-25 21:12:57 +00:00
Benjamin Wright
af1f8655b2 Revert "Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC." 
This reverts commit 7276b974b7.

Reason for revert: Changing to a later Chrome release.

Original change's description:
> Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC.
>
> This change disables DTLS 1.0, TLS 1.0 and TLS 1.1 in WebRTC by default. This
> is part of a larger effort at Google to remove old TLS protocols:
> https://security.googleblog.com/2018/10/modernizing-transport-security.html
>
> For the M74 timeline I have added a disabled by default field trial
> WebRTC-LegacyTlsProtocols which can be enabled to support these cipher suites
> as consumers move away from these legacy cipher protocols but it will be off
> in Chrome.
>
> This is compliant with the webrtc-security-arch specification which states:
>
>    All Implementations MUST implement DTLS 1.2 with the
>    TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the P-256
>    curve [FIPS186].  Earlier drafts of this specification required DTLS
>    1.0 with the cipher suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, and
>    at the time of this writing some implementations do not support DTLS
>    1.2; endpoints which support only DTLS 1.2 might encounter
>    interoperability issues.  The DTLS-SRTP protection profile
>    SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
>    Implementations MUST favor cipher suites which support (Perfect
>    Forward Secrecy) PFS over non-PFS cipher suites and SHOULD favor AEAD
>    over non-AEAD cipher suites.
>
> Bug: webrtc:10261
> Change-Id: I847c567592911cc437f095376ad67585b4355fc0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125141
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: David Benjamin <davidben@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27006}

TBR=steveanton@webrtc.org,davidben@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10261
Change-Id: I34727e65c069e1fb2ad71838828ad0a22b5fe811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130367
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27403}
 2019-04-01 19:11:07 +00:00
Benjamin Wright
7276b974b7 Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC. 
This change disables DTLS 1.0, TLS 1.0 and TLS 1.1 in WebRTC by default. This
is part of a larger effort at Google to remove old TLS protocols:
https://security.googleblog.com/2018/10/modernizing-transport-security.html

For the M74 timeline I have added a disabled by default field trial
WebRTC-LegacyTlsProtocols which can be enabled to support these cipher suites
as consumers move away from these legacy cipher protocols but it will be off
in Chrome.

This is compliant with the webrtc-security-arch specification which states:

   All Implementations MUST implement DTLS 1.2 with the
   TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the P-256
   curve [FIPS186].  Earlier drafts of this specification required DTLS
   1.0 with the cipher suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, and
   at the time of this writing some implementations do not support DTLS
   1.2; endpoints which support only DTLS 1.2 might encounter
   interoperability issues.  The DTLS-SRTP protection profile
   SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
   Implementations MUST favor cipher suites which support (Perfect
   Forward Secrecy) PFS over non-PFS cipher suites and SHOULD favor AEAD
   over non-AEAD cipher suites.

Bug: webrtc:10261
Change-Id: I847c567592911cc437f095376ad67585b4355fc0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125141
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: David Benjamin <davidben@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27006}
 2019-03-06 20:44:41 +00:00
Benjamin Wright
8e98c60f84 Cleanup for openssl_stream_adapter.cc. 
This is a partial cleanup there is more work to do here. Essentially I am just
moving things from static to anonymous namespaces and reordering things to
make more sense. I have removed some old microsoft compiler warning
supressions which I believe are not required anymore.

After this BIO should be refactored to use proper style.

Bug: webrtc:9860
Change-Id: I8419be002d8f412dd89f37f3b865794792ccf559
Reviewed-on: https://webrtc-review.googlesource.com/c/120863
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26912}
 2019-03-01 02:25:13 +00:00
Benjamin Wright
f54e30b596 Add const to variables in openssl_stream_adapter.cc that can use it. 
This change simply adds const to all the variables that can use it. It seemed
like a good idea to enforce const correctness where possible in the TLS stack.

Bug: webrtc:9860
Change-Id: Iabfe1e26647b0c21e2f209e0e0f96d0ec7465e7a
Reviewed-on: https://webrtc-review.googlesource.com/c/124623
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26865}
 2019-02-27 02:44:09 +00:00
Sergey Sablin
3c119fb793 Handle HKDF key derivation when building with OpenSSL. 
Change-Id: I3fd503109190d6a94e15576312c9cb79906a7f61
Bug: webrtc:10160
Reviewed-on: https://webrtc-review.googlesource.com/c/122622
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26669}
 2019-02-13 17:44:02 +00:00
David Benjamin
170a4b383f Trim unnecessary OpenSSL/BoringSSL ifdefs. 
Now that WebRTC requires OpenSSL 1.1.0 as minimum, some bits can be
removed. The simpler versioning API is shared between BoringSSL and
OpenSSL 1.1.0, and there are some remnants of the threading callbacks
that can be removed.

Bug: none
Change-Id: I2078ca9c444b1f1efa9e4b235eb4e6037865d8fb
Reviewed-on: https://webrtc-review.googlesource.com/c/120261
Commit-Queue: David Benjamin <davidben@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26475}
 2019-01-30 17:09:49 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
 2019-01-11 17:11:39 +00:00
Steve Anton
1c05765831 (3) Rename files to snake_case: move the files 
Mechanically generated with this command:

tools_webrtc/do-rename.sh move all-renames.txt

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26225}
 2019-01-11 17:05:20 +00:00
Renamed from rtc_base/opensslstreamadapter.cc (Browse further)