Commit graph

14 commits

Author SHA1 Message Date
Taylor Brandstetter
c392866d86 Implement certificate chain stats.
There was an implementation, but it relied on SSLCertificate::GetChain,
which was never implemented. Except in the fake certificate classes
used by the stats collector tests, hence the tests were passing.

Instead of implementing GetChain, we decided (in
https://webrtc-review.googlesource.com/c/src/+/6500) to add
methods that return a SSLCertChain directly, since it results in a
somewhat cleaner object model.

So this CL switches everything to use the "chain" methods, and gets
rid of the obsolete methods and member variables.

Bug: webrtc:8920
Change-Id: Ie9d7d53654ba859535462521b54c788adec7badf
Reviewed-on: https://webrtc-review.googlesource.com/56961
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22177}
2018-02-24 00:44:06 +00:00
Jiawei Ou
eb0df088ca Update SSL call sites to compile with both OpenSSL 1.1.0 and BoringSSL
OpenSSL is making a lot of data structure opaque, so we can no longer directly access internal data structure. Fortunately, API methods are provided for this purpose.

BoringSSL is sharing the same API.

Bug: webrtc:8817
Change-Id: Ia5090200f0e7c352f82e8191720ac4c14fbb5a85
Reviewed-on: https://webrtc-review.googlesource.com/47321
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21895}
2018-02-05 16:47:35 +00:00
Jiawei Ou
a9c94d5b12 Be explicit about OpenSSL version requriement.
https://chromium-review.googlesource.com/c/external/webrtc/+/575910 pretty much made it a mandate to have OpenSSL 1.1.0 to compile webrtc.

So, let's be explicit about it and cleanup old code for older version support.

Also, generate a compiler error for older OpenSSL versions.

Bug: webrtc:8817
Change-Id: I28590348137b6a04503eabdcc6328297ecf5213e
Reviewed-on: https://webrtc-review.googlesource.com/46502
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#21861}
2018-02-01 22:21:12 +00:00
Jonas Olsson
addc380168 Change some SSL logging to use DLOG
Bug: webrtc:8529
Change-Id: I0242ff201c5c7ac00169444a346e462157703ac6
Reviewed-on: https://webrtc-review.googlesource.com/46260
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21836}
2018-02-01 09:53:51 +00:00
Jiawei Ou
018dd6e9d1 Refer to the underlying object when reporting the state of SSL basic I/O
The reasons behind this change:

1. In OpenSSL 1.1.0. BIO will be an opaque object. We won't have direct access to the `num` field.
2. `num` is only used by OpenSSL provided BIOs and different types of BIOs use num differently.
WebRTC is providing its own customized BIO implementation, it probably shouldn't piggyback into
this internal field to store the stream/socket state.
4. We can access the stream/socket state directly using the underlying object anyway.


Bug: webrtc:8817
Change-Id: I41cdd2920fba378e312e8436a7b9733381555522
Reviewed-on: https://webrtc-review.googlesource.com/46360
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21814}
2018-01-30 21:07:18 +00:00
David Benjamin
3c1f05db84 Fix comment about cipher names.
The comment is backwards. BoringSSL has SSL_CIPHER_standard_name, but
because it is usually not available in OpenSSL, you have to do it
manually. (The code in question is not compiled in BoringSSL.)

Bug: none
Change-Id: If294937afc75d0b0bd3107fd5c57a85c6252f188
Reviewed-on: https://webrtc-review.googlesource.com/28380
Commit-Queue: David Benjamin <davidben@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21146}
2017-12-07 18:32:58 +00:00
Karl Wiberg
e40468ba3d Move some numeric utility code from rtc_base/ to rtc_base/numerics/
Specifically, I'm moving

  safe_compare.h
  safe_conversions.h
  safe_minmax.h

They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.

BUG=webrtc:8445

Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
2017-11-22 11:21:47 +00:00
Jian Cui
0a8798b266 Implement OpenSSLCertificate::GetChain
Continue from https://codereview.webrtc.org/3010363002/

BUG=webrtc:8289

Change-Id: I8d14ba7974b654387f63ff30dee822b3045edcc7
Reviewed-on: https://webrtc-review.googlesource.com/6500
Commit-Queue: Jian Cui <jiancui@google.com>
Reviewed-by: David Benjamin <davidben@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20723}
2017-11-17 01:50:00 +00:00
Mirko Bonadei
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
David Benjamin
dc24656e5e Only verify the certificate once.
WebRTC is currently using the SSL_CTX_set_verify callback. This
configures a callback for use with X509_STORE_CTX_set_verify_cb. See
https://www.openssl.org/docs/man1.0.2/crypto/X509_STORE_CTX_set_verify_cb.html

This callback does not override certificate verification. Rather, it
allows EACH failure in OpenSSL's built-in certificate verification, as
well as the final success, to be overridden (that's why there's an ok
parameter). It still runs the usual OpenSSL certificate verification
(which will never succeed).

The upshot is that the callback is called multiple times and
OpenSSLStreamAdapter does a ton of redundant work and checks the hash at
least twice, or more for certificates with other errors.

Instead, use SSL_CTX_set_cert_verify_callback. This short-circuits the
OpenSSL behavior entirely and uses a caller-supplied one.
https://commondatastorage.googleapis.com/chromium-boringssl-docs/ssl.h.html#SSL_CTX_set_cert_verify_callback
https://wiki.openssl.org/index.php/Manual:SSL_CTX_set_cert_verify_callback(3)

(This also removes the SSL_CTX_set_verify_depth call which is ignored
with SSL_CTX_set_cert_verify_callback. It didn't do anything before
either---it tells OpenSSL to reject chains that are too short, but the
rejection was overwritten by the callback anyway.)

(Later on, we'll need to switch this to the BoringSSL-only
SSL_CTX_set_custom_verify and CRYPTO_BUFFER APIs to fix WebRTC's
contribution to Chrome's binary size, but I've left that alone for the
time being.)

Bug: none
Change-Id: I9320a367d0961935836df63dc6f0868b069f0af0
Reviewed-on: https://webrtc-review.googlesource.com/4581
Commit-Queue: David Benjamin <davidben@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20053}
2017-10-01 01:13:51 +00:00
David Benjamin
85aa0b62dd Mark methods_stream as const.
Function pointer tables require relocations, so this goes into
.data.rel.ro, not .rodata, but this will at least mark the pages
read-only after relocations are resolved.

Bug: None
Change-Id: I8625e7466b2dcadafc4e4e5f9c6eccbd87af7109
Reviewed-on: https://webrtc-review.googlesource.com/4580
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20029}
2017-09-29 00:58:07 +00:00
David Benjamin
a8f7376789 Switch from SSL_CIPHER_get_rfc_name to SSL_CIPHER_standard_name.
SSL_CIPHER_standard_name is a bit easier to use. BoringSSL has the
strings in the library statically these days. (Turns out that's more
size-efficient than the code to build it up anyway!)

Bug: None
Change-Id: I91ffa725fa716791cdf75d944cf8d9a3e2cb9021
Reviewed-on: https://webrtc-review.googlesource.com/4362
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20028}
2017-09-29 00:56:56 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/rtc_base/opensslstreamadapter.cc (Browse further)