Commit graph

10 commits

Author SHA1 Message Date
Sam Zackrisson
b298f743b8 Revert "Replace sigslot usages with robocaller library."
This reverts commit 40261c3663.

Reason for revert: Breaks downstream project

Original change's description:
> Replace sigslot usages with robocaller library.
>
> - Replace all the top level signals from jsep_transport_controller.
> - There are still sigslot usages in this file so keep the inheritance
>   and that is the reason for not having a binary size gain in this CL.
>
> Bug: webrtc:11943
> Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
> Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32321}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,glahiru@webrtc.org

Change-Id: Icf438f87c3d95940d858db3cc5848b23abb82fc4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186844
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32324}
2020-10-06 11:40:43 +00:00
Lahiru Ginnaliya Gamathige
40261c3663 Replace sigslot usages with robocaller library.
- Replace all the top level signals from jsep_transport_controller.
- There are still sigslot usages in this file so keep the inheritance
  and that is the reason for not having a binary size gain in this CL.

Bug: webrtc:11943
Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32321}
2020-10-05 22:38:57 +00:00
Harald Alvestrand
b33a0ca1ee Remove deprecated ssl_identity methods
This is a followup to
https://webrtc-review.googlesource.com/c/src/+/170637

Bug: webrtc:11450
Change-Id: I69928ed7236c6a8a569c7dc0383f7debb4408179
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171224
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31086}
2020-04-16 14:21:41 +00:00
Harald Alvestrand
8515d5a4ab Refactor ssl_stream_adapter API to show object ownership
Backwards compatible overloads are provided.

Bug: none
Change-Id: I065ad6b269fe074745f9debf68862ff70fd09628
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170637
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30851}
2020-03-21 18:53:46 +00:00
Harald Alvestrand
137991396d Make a switch to disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC.
This reverts commit af1f8655b2

Landing the change with default set to
"enabled" (DTLS 1.0 will continue to work by default),
so that flipping the default can be a separate CL.

Original change's description:
> 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}

Bug: webrtc:10261
Change-Id: I28c6819d37665976e396df280b4abf48fb91d533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169851
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30733}
2020-03-09 19:23:44 +00:00
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
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
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/sslstreamadapter.h (Browse further)