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Saúl Ibarra Corretgé d8633868b3 Enable SRTP GCM ciphers by default 
Bug: webrtc:15178
Change-Id: I0216ce8f194fffc820723d82b9c04a76573c2f4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305381
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40828}
2023-09-28 10:18:56 +00:00
..
g3doc
Document how bitrate probing works from a RTP perspective
2023-06-09 14:14:12 +00:00
scenario_tests
…
test
Fire MaybeSignalReadyToSend in a PostTask when recursive
2023-09-27 07:36:40 +00:00
audio_rtp_receiver_unittest.cc
Remove all split channels related code
2023-06-29 09:32:04 +00:00
audio_rtp_receiver.cc
Rename Set(Send|Recv)Parameters Set(Sender|Receiver)Parameters
2023-09-01 08:12:55 +00:00
audio_rtp_receiver.h
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audio_track.cc
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audio_track.h
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BUILD.gn
Fire MaybeSignalReadyToSend in a PostTask when recursive
2023-09-27 07:36:40 +00:00
channel_interface.h
Remove all split channels related code
2023-06-29 09:32:04 +00:00
channel_unittest.cc
Look through all candidates before falling back to default packetization
2023-09-20 12:18:02 +00:00
channel.cc
Look through all candidates before falling back to default packetization
2023-09-20 12:18:02 +00:00
channel.h
Convert signals in rtp_transport_internal.h to CallbackList
2023-09-05 11:37:32 +00:00
connection_context.cc
Reland "ConnectionContext: remove media engine without blocking."
2023-09-06 08:20:44 +00:00
connection_context.h
Reland "ConnectionContext: remove media engine without blocking."
2023-09-06 08:20:44 +00:00
data_channel_controller_unittest.cc
Attempt to recycle a stopped data m-line before creating a new one
2023-05-09 15:11:24 +00:00
data_channel_controller.cc
Avoid touching channel after OnSctpDataChannelClosed
2023-06-22 09:13:29 +00:00
data_channel_controller.h
Refactor HasDataChannels
2023-05-15 14:52:12 +00:00
data_channel_integrationtest.cc
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data_channel_unittest.cc
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data_channel_utils.cc
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data_channel_utils.h
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
DEPS
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dtls_srtp_transport_unittest.cc
Convert signals in rtp_transport_internal.h to CallbackList
2023-09-05 11:37:32 +00:00
dtls_srtp_transport.cc
Reduce logging verbosity of DTLS-SRTP RTCP transport
2023-08-22 13:48:09 +00:00
dtls_srtp_transport.h
More unused sigslot includes
2023-08-29 12:20:44 +00:00
dtls_transport_unittest.cc
…
dtls_transport.cc
…
dtls_transport.h
…
dtmf_sender_unittest.cc
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dtmf_sender.cc
…
dtmf_sender.h
…
external_hmac.cc
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external_hmac.h
…
ice_server_parsing_unittest.cc
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ice_server_parsing.cc
…
ice_server_parsing.h
…
ice_transport_unittest.cc
…
ice_transport.cc
…
ice_transport.h
…
jitter_buffer_delay_unittest.cc
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jitter_buffer_delay.cc
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jitter_buffer_delay.h
…
jsep_ice_candidate.cc
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jsep_session_description_unittest.cc
Remove references to AudioCodec and VideoCodec constructors
2023-06-05 23:23:40 +00:00
jsep_session_description.cc
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jsep_transport_collection.cc
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jsep_transport_collection.h
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jsep_transport_controller_unittest.cc
Allow single-mline offers without BUNDLE group when using max-bundle
2023-05-26 06:10:50 +00:00
jsep_transport_controller.cc
Fix DCHECK crash when processing a remote answer
2023-09-08 10:08:30 +00:00
jsep_transport_controller.h
…
jsep_transport_unittest.cc
…
jsep_transport.cc
Rename cipher_suite to crypto_suite
2023-08-02 11:45:24 +00:00
jsep_transport.h
…
legacy_stats_collector_interface.h
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legacy_stats_collector_unittest.cc
Replace "rcvd" with "received" for readability
2023-04-24 15:30:07 +00:00
legacy_stats_collector.cc
Remove default "unknown" encoderImplementation/decoderImplementation
2023-06-22 11:49:58 +00:00
legacy_stats_collector.h
…
local_audio_source_unittest.cc
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local_audio_source.cc
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local_audio_source.h
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media_protocol_names.cc
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media_protocol_names.h
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media_session_unittest.cc
Enable SRTP GCM ciphers by default
2023-09-28 10:18:56 +00:00
media_session.cc
MediaSession: ensure transport description factory exists
2023-09-28 08:26:05 +00:00
media_session.h
MediaSession: ensure transport description factory exists
2023-09-28 08:26:05 +00:00
media_stream_observer.cc
…
media_stream_observer.h
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media_stream_proxy.h
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media_stream_track_proxy.h
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media_stream_unittest.cc
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media_stream.cc
…
media_stream.h
…
OWNERS
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peer_connection_adaptation_integrationtest.cc
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peer_connection_bundle_unittest.cc
Remove another ctor from BasicPortAllocator
2023-09-01 11:55:43 +00:00
peer_connection_crypto_unittest.cc
Rename cipher_suite to crypto_suite
2023-08-02 11:45:24 +00:00
peer_connection_data_channel_unittest.cc
…
peer_connection_encodings_integrationtest.cc
Implement codec selection api
2023-08-24 13:18:04 +00:00
peer_connection_end_to_end_unittest.cc
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peer_connection_factory_proxy.h
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peer_connection_factory_unittest.cc
Check use_rtx() in PeerConnectionFactory::GetRtpSenderCapabilities
2023-08-25 09:08:32 +00:00
peer_connection_factory.cc
Deprecate AsyncResolver config fields and remove internal usage.
2023-08-25 14:02:27 +00:00
peer_connection_factory.h
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peer_connection_field_trial_tests.cc
Reland: FrameGeneratorCapturer: don't generate video before Start is called
2023-06-29 14:47:05 +00:00
peer_connection_header_extension_unittest.cc
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peer_connection_histogram_unittest.cc
Remove another ctor from BasicPortAllocator
2023-09-01 11:55:43 +00:00
peer_connection_ice_unittest.cc
Remove another ctor from BasicPortAllocator
2023-09-01 11:55:43 +00:00
peer_connection_integrationtest.cc
Deprecate AsyncResolver config fields and remove internal usage.
2023-08-25 14:02:27 +00:00
peer_connection_interface_unittest.cc
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peer_connection_internal.h
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peer_connection_jsep_unittest.cc
sdp: accept bundle-only media section without rtcp-mux
2023-05-12 13:38:23 +00:00
peer_connection_media_unittest.cc
Implement codec selection api
2023-08-24 13:18:04 +00:00
peer_connection_message_handler.cc
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peer_connection_message_handler.h
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peer_connection_proxy.h
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peer_connection_rampup_tests.cc
Remove another ctor from BasicPortAllocator
2023-09-01 11:55:43 +00:00
peer_connection_rtp_unittest.cc
sdp: add validation for the number of ssrcs in the ssrc group
2023-08-29 11:33:51 +00:00
peer_connection_signaling_unittest.cc
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peer_connection_simulcast_unittest.cc
Issue 1447193: Remove expired histograms WebRTC.PeerConnection.Simulcast.ApplyLocalDescription, WebRTC.PeerConnection.Simulcast.ApplyRemoteDescription, WebRTC.PeerConnection.Simulcast.Disabled
2023-07-12 06:50:15 +00:00
peer_connection_svc_integrationtest.cc
Implement codec selection api
2023-08-24 13:18:04 +00:00
peer_connection_wrapper.cc
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peer_connection_wrapper.h
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peer_connection.cc
Deprecate AsyncResolver config fields and remove internal usage.
2023-08-25 14:02:27 +00:00
peer_connection.h
Delete some unused sigslot includes
2023-08-28 12:36:39 +00:00
proxy_unittest.cc
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proxy.cc
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proxy.h
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
remote_audio_source.cc
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remote_audio_source.h
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rtc_stats_collector_unittest.cc
stats: implement fecSsrc on inbound-rtp
2023-08-04 12:54:48 +00:00
rtc_stats_collector.cc
stats: implement fecSsrc on inbound-rtp
2023-08-04 12:54:48 +00:00
rtc_stats_collector.h
Delete deprecated 'track' and 'stream' metrics from WebRTC.
2023-05-24 12:26:56 +00:00
rtc_stats_integrationtest.cc
stats: implement fecSsrc on inbound-rtp
2023-08-04 12:54:48 +00:00
rtc_stats_traversal_unittest.cc
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rtc_stats_traversal.cc
Delete deprecated 'track' and 'stream' metrics from WebRTC.
2023-05-24 12:26:56 +00:00
rtc_stats_traversal.h
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rtcp_mux_filter_unittest.cc
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rtcp_mux_filter.cc
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rtcp_mux_filter.h
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rtp_media_utils_unittest.cc
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rtp_media_utils.cc
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rtp_media_utils.h
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rtp_parameters_conversion_unittest.cc
Reland "Merge the codec types"
2023-06-15 15:53:29 +00:00
rtp_parameters_conversion.cc
Reland "Merge the codec types"
2023-06-15 15:53:29 +00:00
rtp_parameters_conversion.h
Reland "Merge the codec types"
2023-06-15 15:53:29 +00:00
rtp_receiver_proxy.h
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rtp_receiver.cc
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rtp_receiver.h
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rtp_sender_proxy.h
Remove public GenerateKeyFrame(list-of-rids) API from RtpSender
2023-08-30 14:54:17 +00:00
rtp_sender_receiver_unittest.cc
Rename Set(Send|Recv)Parameters Set(Sender|Receiver)Parameters
2023-09-01 08:12:55 +00:00
rtp_sender.cc
Implement codec selection api
2023-08-24 13:18:04 +00:00
rtp_sender.h
Remove public GenerateKeyFrame(list-of-rids) API from RtpSender
2023-08-30 14:54:17 +00:00
rtp_transceiver_unittest.cc
Remove all split channels related code
2023-06-29 09:32:04 +00:00
rtp_transceiver.cc
Implement codec selection api
2023-08-24 13:18:04 +00:00
rtp_transceiver.h
Change SSRC-passing for MediaChannel from external to callback
2023-05-22 14:33:59 +00:00
rtp_transmission_manager.cc
…
rtp_transmission_manager.h
Delete some unused sigslot includes
2023-08-28 12:36:39 +00:00
rtp_transport_internal.h
Convert signals in rtp_transport_internal.h to CallbackList
2023-09-05 11:37:32 +00:00
rtp_transport_unittest.cc
Fire MaybeSignalReadyToSend in a PostTask when recursive
2023-09-27 07:36:40 +00:00
rtp_transport.cc
Fire MaybeSignalReadyToSend in a PostTask when recursive
2023-09-27 07:36:40 +00:00
rtp_transport.h
Fire MaybeSignalReadyToSend in a PostTask when recursive
2023-09-27 07:36:40 +00:00
sctp_data_channel.cc
[SctpDataChannel] Don't use PostTask for observer registration.
2023-08-25 09:07:29 +00:00
sctp_data_channel.h
…
sctp_transport_unittest.cc
…
sctp_transport.cc
…
sctp_transport.h
…
sctp_utils_unittest.cc
…
sctp_utils.cc
…
sctp_utils.h
…
sdp_offer_answer_unittest.cc
Reject attempts to change the media kind for a m-line with a previously used mid
2023-09-20 12:21:24 +00:00
sdp_offer_answer.cc
Unify RTP payload type validity checking
2023-09-25 14:54:22 +00:00
sdp_offer_answer.h
…
sdp_state_provider.h
…
sdp_utils.cc
…
sdp_utils.h
…
session_description_unittest.cc
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session_description.cc
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
session_description.h
Move codecs() to MediaContentDescription
2023-09-20 10:16:36 +00:00
simulcast_description.cc
…
simulcast_description.h
…
simulcast_sdp_serializer_unittest.cc
Rename simulcast SDP serializer
2023-09-18 10:09:02 +00:00
simulcast_sdp_serializer.cc
Rename simulcast SDP serializer
2023-09-18 10:09:02 +00:00
simulcast_sdp_serializer.h
Rename simulcast SDP serializer
2023-09-18 10:09:02 +00:00
slow_peer_connection_integration_test.cc
[SourceTracker] Move state to the worker thread, remove mutex.
2023-04-25 08:18:42 +00:00
srtp_filter_unittest.cc
Rename cipher_suite to crypto_suite
2023-08-02 11:45:24 +00:00
srtp_filter.cc
Rename cipher_suite to crypto_suite
2023-08-02 11:45:24 +00:00
srtp_filter.h
Rename cipher_suite to crypto_suite
2023-08-02 11:45:24 +00:00
srtp_session_unittest.cc
…
srtp_session.cc
Rename cipher_suite to crypto_suite
2023-08-02 11:45:24 +00:00
srtp_session.h
Rename cipher_suite to crypto_suite
2023-08-02 11:45:24 +00:00
srtp_transport_unittest.cc
Convert signals in rtp_transport_internal.h to CallbackList
2023-09-05 11:37:32 +00:00
srtp_transport.cc
Convert signals in rtp_transport_internal.h to CallbackList
2023-09-05 11:37:32 +00:00
srtp_transport.h
Rename cipher_suite to crypto_suite
2023-08-02 11:45:24 +00:00
stream_collection.h
…
track_media_info_map_unittest.cc
Delete deprecated 'track' and 'stream' metrics from WebRTC.
2023-05-24 12:26:56 +00:00
track_media_info_map.cc
Delete deprecated 'track' and 'stream' metrics from WebRTC.
2023-05-24 12:26:56 +00:00
track_media_info_map.h
Delete deprecated 'track' and 'stream' metrics from WebRTC.
2023-05-24 12:26:56 +00:00
transceiver_list.cc
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transceiver_list.h
…
transport_stats.cc
…
transport_stats.h
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usage_pattern.cc
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usage_pattern.h
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used_ids_unittest.cc
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used_ids.h
Unify RTP payload type validity checking
2023-09-25 14:54:22 +00:00
video_rtp_receiver_unittest.cc
Split fake media channel classes
2023-06-28 11:40:47 +00:00
video_rtp_receiver.cc
Rename Set(Send|Recv)Parameters Set(Sender|Receiver)Parameters
2023-09-01 08:12:55 +00:00
video_rtp_receiver.h
…
video_rtp_track_source_unittest.cc
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video_rtp_track_source.cc
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video_rtp_track_source.h
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video_track_source_proxy.cc
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video_track_source_proxy.h
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video_track_source.cc
…
video_track_source.h
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video_track_unittest.cc
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video_track.cc
…
video_track.h
…
webrtc_sdp_unittest.cc
sdp: reject duplicate ssrcs in ssrc-groups
2023-06-15 14:24:54 +00:00
webrtc_sdp.cc
Move codecs() to MediaContentDescription
2023-09-20 10:16:36 +00:00
webrtc_sdp.h
…
webrtc_session_description_factory.cc
Make CreateOffer/CreateAnswer return RTCErrorOr<SessionDescription>
2023-09-26 09:41:30 +00:00
webrtc_session_description_factory.h
Use RTCError instead of string for PostCreateSessionDescriptionFailed
2023-09-18 15:23:38 +00:00