| f475d365a25036725c3f545f57de59d2cc902d17 |
|
09-Jan-2016 |
Taylor Brandstetter <deadbeef@webrtc.org> |
Properly handle different transports having different SSL roles.
This meant splitting "transport_options" into audio/video/data options,
for when creating the answer, and giving "GetSslRole" a "transport_name"
parameter so we can retrieve the current role on a per-transport basis.
BUG=webrtc:4525
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1516993002 .
Cr-Commit-Position: refs/heads/master@{#11192}
/external/webrtc/talk/session/media/mediasession.h
|
| 1387149ad1669365ac05278bf779a407bec08a4e |
|
09-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding reduced size RTCP configuration down to the video stream level.
Still waiting to turn on negotiation (in mediasession.cc)
until we verify it's working as expected.
BUG=webrtc:4868
Review URL: https://codereview.webrtc.org/1418123003
Cr-Commit-Position: refs/heads/master@{#10958}
/external/webrtc/talk/session/media/mediasession.h
|
| 521ed7bf022c4e30574d7970c2be5be46567f4cd |
|
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/mediasession.h
|
| 318166bed75dcbc00a7b79f715f9953aff9ffbc7 |
|
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/mediasession.h
|
| 2764e1027a08a5543e04b854a27a520801faf6eb |
|
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/mediasession.h
|
| 0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
|
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/mediasession.h
|
| 456696a9c1bbd586701dcca3e4b2695e419a10ba |
|
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/mediasession.h
|
| 27dc29b0df23eed5034f28d4d5f66ea0bb425d6c |
|
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/mediasession.h
|
| 4fe3c9a77386598db9abd1f0d6983aefee9cc943 |
|
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/mediasession.h
|
| 7cbd188c5ed7df80bb737bd4ada94422730e2d89 |
|
18-Sep-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GICE (again).
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1353713002 .
Cr-Commit-Position: refs/heads/master@{#9979}
/external/webrtc/talk/session/media/mediasession.h
|
| d12140a68efdcffa1c2c18f25149905e9dae1a9c |
|
10-Sep-2015 |
guoweis <guoweis@webrtc.org> |
Revert change which removes GICE.
There are still dependencies on this functionality.
TBR=pthatcher@webrtc.org
BUG=526399
Review URL: https://codereview.webrtc.org/1336553003
Cr-Commit-Position: refs/heads/master@{#9920}
/external/webrtc/talk/session/media/mediasession.h
|
| 2159b89fa2cb55beeef38f72bd45e217f3d33d4e |
|
22-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
This reverts commit 5bdafd44c86ee46bd7e040f19828324583418b33.
Original CL: https://codereview.webrtc.org/1263663002/
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1303393002 .
Cr-Commit-Position: refs/heads/master@{#9761}
/external/webrtc/talk/session/media/mediasession.h
|
| 5bdafd44c86ee46bd7e040f19828324583418b33 |
|
21-Aug-2015 |
minyuel <minyue@webrtc.org> |
Revert "Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.""
This reverts commit 081f34b564e1a26ffbbe9515eba1fef7c736fdde.
Original code review see
https://codereview.webrtc.org/1291363005
The revert is due to a suspicion of "Reland "Remove GICE..." being the cause of failure on Linux memcheck, see
https://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/4137
TBR=pthatcher@webrtc.org,
BUG=
Review URL: https://codereview.webrtc.org/1308753003 .
Cr-Commit-Position: refs/heads/master@{#9756}
/external/webrtc/talk/session/media/mediasession.h
|
| 081f34b564e1a26ffbbe9515eba1fef7c736fdde |
|
20-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."
This reverts commit 475243a134be003aab30bb17294ca6c664d0ef81.
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1291363005 .
Cr-Commit-Position: refs/heads/master@{#9738}
/external/webrtc/talk/session/media/mediasession.h
|
| fa301809b698017455847f45cc7e0dfa1bdfed35 |
|
11-Aug-2015 |
pthatcher <pthatcher@webrtc.org> |
Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
This reverts commit 3449faa553ec94c52ef2d0949867befb60992c88.
TBR=deadbeef@webrtc.org, juberti@webrtc.org
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1274273005
Cr-Commit-Position: refs/heads/master@{#9698}
/external/webrtc/talk/session/media/mediasession.h
|
| 3449faa553ec94c52ef2d0949867befb60992c88 |
|
10-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever).
R=deadbeef@webrtc.org, juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1263663002 .
Cr-Commit-Position: refs/heads/master@{#9692}
/external/webrtc/talk/session/media/mediasession.h
|
| a74709333482783cb06405626caf9555e407eba2 |
|
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
After another round of reviews.
Cr-Commit-Position: refs/heads/master@{#8483}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8483 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.h
|
| ec97c6516f0f2540a8e040d08de86709be6ab5b4 |
|
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Attempt on read-only acceptance of -12.
Cr-Commit-Position: refs/heads/master@{#8477}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8477 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.h
|
| 586f2eda0d90b84ffefdf2c3662073f22af73bdb |
|
23-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Change GetStreamBySsrc to not copy StreamParams.
This is something I stumbled upon while looking at string copying we do (in spades) and did a simple change to not be constantly copying things around needlessly. There's a lot more that can be done in these files of course so this is sort of a reminder for future code edits that it's possible to design interfaces/function in a way that's more performance aware and avoid forcing creation of copies, while still being very simple. Also, we can use lambdas now :)
BUG=
R=perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8131 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.h
|
| 269fb4bc90b79bebbb8311da0110ccd6803fd0a8 |
|
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/mediasession.h
|
| 28100cb38896fe298b6df11ffd31838d9faf5b8a |
|
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/mediasession.h
|
| d1ba6d9cbfc44618d2c553ff7851948c730ae37b |
|
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/mediasession.h
|
| 742922b313baaebfbacf735287f9729a8bc6f8e0 |
|
07-Oct-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Make the media content send only if offerToReceive is false while local streams exist.
We previously do not add the media content if offerToReceive is false.
BUG=3833
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7390 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.h
|
| a09a99950ec40aef6421e4ba35eee7196b7a6e68 |
|
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/mediasession.h
|
| e7d47a1473e885a57986dcdbf06e7e1d25226ca6 |
|
05-Aug-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Maintain the order of the m-lines in CreateOffer and CreateAnswer.
The order in the offer follows the order in the current local description.
The order in the answer follows the order in the current offer.
BUG=2395
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6828 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.h
|
| d4e598d57aed714a599444a7eab5e8fdde52a950 |
|
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/mediasession.h
|
| 8dcd43c4f71da88f75ca46ed5868eb8812e1d6f7 |
|
30-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF.
This is the first step toward switching completely to UDP/TLS/RTP/SAVPF.
BUG=2796
R=juberti@webrtc.org, pthatcher@google.com
Review URL: https://webrtc-codereview.appspot.com/13439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6276 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.h
|
| 9c16c39e613ebc5cdfa8ca5818a62ef5c3b18bd7 |
|
01-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Sets the SCTP port codec in the native SessionDescription.
Previously it's only set when a SDP string is parsed into SessionDescription, causing failuring for native client.
BUG=3141
R=juberti@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6036 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.h
|
| b90991dade9139e5c14c3b616a9eff07b9d6fdda |
|
04-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle 62472237->62550414
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5640 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.h
|
| 4b26e2eee3e3b2a0c22946372a38f7efa6cee146 |
|
16-Jan-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 59676287
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.h
|
| cecfd1832dc375225da3f5f18ecac63006ed06bf |
|
30-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 55821645.
TEST=try bots
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5053 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.h
|
| 1112c30e1e5f5c7b4b517c4954ef3f15b989a996 |
|
23-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 53057474.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2274004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4818 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/session/media/mediasession.h
|
| 28e20752806a492f5a6a5d343c02f9556f39b1cd |
|
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/mediasession.h
|