1218d7ad2fac035376914bd0649fe99e657b33d3 |
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05-Dec-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Allow remote fingerprint update during a call Changes include the following 1. modify FakeDtlsIdentityStore to support alternate certificate so we could have a different fingerprint in test case. 2. dtlstransportchannel can accept a new fingerprint and trigger DTLS handshake. 3. #2 will trigger new signal on the media side to reset SRTP context. Only reset SRTP context when we are using DTLS (not SDES). 4. Test cases for caller or callee are transfees. TBR=pthatcher@webrtc.org BUG=webrtc:3618 This is a reland of https://codereview.webrtc.org/1453523002 Review URL: https://codereview.webrtc.org/1505573002 . Cr-Commit-Position: refs/heads/master@{#10903}
/external/webrtc/talk/session/media/srtpfilter.h
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86aaa4be8de8f49f91faeefbfd1a23f312898dd2 |
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05-Dec-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Revert "Allow remote fingerprint update during a call" This reverts commit 9c38c2d33fa6d794704d53b18f39d5235439fe63. This commit somehow is different from what I have in my local copy. Revert and will recommit. TBR=pthatcher@webrtc.org BUG=3618 Review URL: https://codereview.webrtc.org/1494373004 . Cr-Commit-Position: refs/heads/master@{#10902}
/external/webrtc/talk/session/media/srtpfilter.h
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9c38c2d33fa6d794704d53b18f39d5235439fe63 |
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05-Dec-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Allow remote fingerprint update during a call Changes include the following 1. modify FakeDtlsIdentityStore to support alternate certificate so we could have a different fingerprint in test case. 2. dtlstransportchannel can accept a new fingerprint and trigger DTLS handshake. 3. #2 will trigger new signal on the media side to reset SRTP context. Only reset SRTP context when we are using DTLS (not SDES). 4. Test cases for caller or callee are transfees. BUG=webrtc:3618 R=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1453523002 . Cr-Commit-Position: refs/heads/master@{#10901}
/external/webrtc/talk/session/media/srtpfilter.h
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521ed7bf022c4e30574d7970c2be5be46567f4cd |
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19-Nov-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Reland Convert internal representation of Srtp cryptos from string to int TBR=pthatcher@webrtc.org BUG=webrtc:5043 Review URL: https://codereview.webrtc.org/1458023002 . Cr-Commit-Position: refs/heads/master@{#10703}
/external/webrtc/talk/session/media/srtpfilter.h
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318166bed75dcbc00a7b79f715f9953aff9ffbc7 |
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19-Nov-2015 |
guoweis <guoweis@webrtc.org> |
Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ ) Reason for revert: Broke chromium fyi build. Original issue's description: > Convert internal representation of Srtp cryptos from string to int. > > Note that the coversion from int to string happens in 3 places > 1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames. > 2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names. > 3) stats collection also needs external names. > > External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc. > Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc. > > The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams(). > > BUG=webrtc:5043 > > Committed: https://crrev.com/2764e1027a08a5543e04b854a27a520801faf6eb > Cr-Commit-Position: refs/heads/master@{#10701} TBR=juberti@webrtc.org,pthatcher@webrtc.org,juberti@google.com NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5043 Review URL: https://codereview.webrtc.org/1455233005 Cr-Commit-Position: refs/heads/master@{#10702}
/external/webrtc/talk/session/media/srtpfilter.h
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2764e1027a08a5543e04b854a27a520801faf6eb |
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19-Nov-2015 |
guoweis <guoweis@webrtc.org> |
Convert internal representation of Srtp cryptos from string to int. Note that the coversion from int to string happens in 3 places 1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames. 2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names. 3) stats collection also needs external names. External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc. Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc. The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams(). BUG=webrtc:5043 Review URL: https://codereview.webrtc.org/1416673006 Cr-Commit-Position: refs/heads/master@{#10701}
/external/webrtc/talk/session/media/srtpfilter.h
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cbe9f51cf85a5aeb20a5134dad56cd2b527c098d |
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13-Nov-2015 |
phoglund <phoglund@webrtc.org> |
Revert of Remove global list of SRTP sessions. (patchset #4 id:60001 of https://codereview.webrtc.org/1416093010/ ) Reason for revert: Unfortunately this breaks an internal downstream project since we have an ancient libsrtp. Reverting until we can figure out how to update our libsrtp. Original issue's description: > Remove global list of SRTP sessions. > Instead save a reference to the SrtpSession inside the srtp_ctx_t. > > BUG=webrtc:5133 > > Committed: https://crrev.com/9cafd972779ed7b25886ab276e0ede7b7a8b76a1 > Cr-Commit-Position: refs/heads/master@{#10591} TBR=juberti@google.com,juberti@webrtc.org,jbauch@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5133 Review URL: https://codereview.webrtc.org/1442863003 Cr-Commit-Position: refs/heads/master@{#10635}
/external/webrtc/talk/session/media/srtpfilter.h
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9cafd972779ed7b25886ab276e0ede7b7a8b76a1 |
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10-Nov-2015 |
jbauch <jbauch@webrtc.org> |
Remove global list of SRTP sessions. Instead save a reference to the SrtpSession inside the srtp_ctx_t. BUG=webrtc:5133 Review URL: https://codereview.webrtc.org/1416093010 Cr-Commit-Position: refs/heads/master@{#10591}
/external/webrtc/talk/session/media/srtpfilter.h
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0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
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07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types. Removes the use of uint8, etc. in favor of uint8_t. BUG=webrtc:5024 R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1362503003 . Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/talk/session/media/srtpfilter.h
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456696a9c1bbd586701dcca3e4b2695e419a10ba |
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01-Oct-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Reland Change WebRTC SslCipher to be exposed as number only This is to revert the change of https://codereview.webrtc.org/1380603005/ TBR=pthatcher@webrtc.org BUG=523033 Review URL: https://codereview.webrtc.org/1375543003 . Cr-Commit-Position: refs/heads/master@{#10126}
/external/webrtc/talk/session/media/srtpfilter.h
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27dc29b0df23eed5034f28d4d5f66ea0bb425d6c |
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01-Oct-2015 |
guoweis <guoweis@webrtc.org> |
Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ ) Reason for revert: This broke chromium.fyi bot. Original issue's description: > Change WebRTC SslCipher to be exposed as number only. > > This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting. > > For SRTP, currently it's still string internally but is reported as IANA number. > > This is used by the ongoing CL https://codereview.chromium.org/1335023002. > > BUG=523033 > > Committed: https://crrev.com/4fe3c9a77386598db9abd1f0d6983aefee9cc943 > Cr-Commit-Position: refs/heads/master@{#10124} TBR=juberti@webrtc.org,rsleevi@chromium.org,pthatcher@webrtc.org,davidben@chromium.org,juberti@google.com,davidben@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=523033 Review URL: https://codereview.webrtc.org/1380603005 Cr-Commit-Position: refs/heads/master@{#10125}
/external/webrtc/talk/session/media/srtpfilter.h
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4fe3c9a77386598db9abd1f0d6983aefee9cc943 |
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01-Oct-2015 |
guoweis <guoweis@webrtc.org> |
Change WebRTC SslCipher to be exposed as number only. This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting. For SRTP, currently it's still string internally but is reported as IANA number. This is used by the ongoing CL https://codereview.chromium.org/1335023002. BUG=523033 Review URL: https://codereview.webrtc.org/1337673002 Cr-Commit-Position: refs/heads/master@{#10124}
/external/webrtc/talk/session/media/srtpfilter.h
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3c089d751ede283e21e186885eaf705c3257ccd2 |
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16-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to contructormagic macros. We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition. * DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN * DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN * DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS Related CL: https://codereview.webrtc.org/1335923002/ BUG=chromium:468375 NOTRY=true Review URL: https://codereview.webrtc.org/1345433002 Cr-Commit-Position: refs/heads/master@{#9953}
/external/webrtc/talk/session/media/srtpfilter.h
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fec2c6d7eb58574b32eaa26222d3fb903b738cfa |
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27-May-2015 |
Joachim Bauch <jbauch@webrtc.org> |
Prevent potential double-free if srtp_create fails. If srtp_create fails while adding streams, it deallocates the session but doesn't clear the passed pointer which then could lead to a double-free in the SrtpSession dtor. The CL also adds locking for libsrtp initialization / shutdown. BUG=4042 R=jiayl@webrtc.org, juberti@google.com, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/47319004 Cr-Commit-Position: refs/heads/master@{#9300}
/external/webrtc/talk/session/media/srtpfilter.h
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e728ee03ba093ddb9fa6fb803994969801a4f601 |
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17-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove or rename typedefs with _t prefixes. _t prefixes are reserved for additional typenames in POSIX. R=henrik.lundin@webrtc.org, hta@webrtc.org, stefan@webrtc.org BUG=162 Review URL: https://webrtc-codereview.appspot.com/36559004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7931 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/srtpfilter.h
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269fb4bc90b79bebbb8311da0110ccd6803fd0a8 |
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28-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
move xmpp and p2p to webrtc Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder. BUG=3379 Review URL: https://webrtc-codereview.appspot.com/26999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/srtpfilter.h
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28100cb38896fe298b6df11ffd31838d9faf5b8a |
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18-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." BUG=N/A TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/srtpfilter.h
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d1ba6d9cbfc44618d2c553ff7851948c730ae37b |
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15-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. BUG=3379 R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27709005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/srtpfilter.h
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a09a99950ec40aef6421e4ba35eee7196b7a6e68 |
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13-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73222930-> 73226398 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/srtpfilter.h
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d4e598d57aed714a599444a7eab5e8fdde52a950 |
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29-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72097588-> 72159069 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/srtpfilter.h
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a7b981843f35bb6c26cf3bc95b5a00a0b9f50a93 |
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21-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702). BUG=N/A R=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5595 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/srtpfilter.h
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ef2215110c00ee1d8225b08815bfdcee918767f9 |
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21-Feb-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5590 "description" > description TBR=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8949006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5593 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/srtpfilter.h
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2643805a2057b92e916bcf4f71668bc80766625e |
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20-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
description git-svn-id: http://webrtc.googlecode.com/svn/trunk@5590 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/srtpfilter.h
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28ff3ee6aa34d1386f61c9277feaa41ec8c919ee |
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16-Aug-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix invalid cricket::SrtpStat::FailureKey::operator<() implementation. If operator<(a, b) returns true, then it must not be the case that operator<(b, a) is true as well, but the old implementation would do exactly that if a={1, 0, 0} and b={0, 0, 1}, for example. Should fix e.g.: [004:555] Error(unittest_main.cc:40): c:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\xtree(1746) : Assertion failed: invalid operator< from http://chromegw/i/client.libjingle/builders/Win32%20Debug/builds/245/steps/libjingle_p2p_unittest/logs/stdio R=juberti@webrtc.org, mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2054005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4561 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/srtpfilter.h
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9dba52562725dbaced0d671982201ede753d72e8 |
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05-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
* Update libjingle to 50389769. * Together with "Add texture support for i420 video frame." from wuchengli@chromium.org. https://webrtc-codereview.appspot.com/1413004 RISK=P1 TESTED=try bots R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1967004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/srtpfilter.h
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28e20752806a492f5a6a5d343c02f9556f39b1cd |
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10-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/srtpfilter.h
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