ac2f3d14e4
Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class that only handles SRTP configuration to a more generic structure that can be used and extended for all per peer connection CryptoOptions that can be on a given PeerConnection. Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be accessed as crypto_options.srtp.whatever_option_name. This is more inline with other structures we have in WebRTC such as VideoConfig. As additional features are added over time this will allow the structure to remain compartmentalized and concerned components can only request a subset of the overall configuration structure e.g: void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); In addition to this it made little sense for sslstreamadapter.h to hold all Srtp related configuration options. The header has become loo large and takes on too many responsibilities and spilting this up will lead to more maintainable code going forward. This will be used in a future CL to enable configuration options for the newly supported Frame Crypto. Change-Id: I99d1be36740c59548c8e62db52d68d738649707f Bug: webrtc:9681 Reviewed-on: https://webrtc-review.googlesource.com/c/105180 Reviewed-by: Emad Omara <emadomara@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25130}
53 lines
1.7 KiB
C++
53 lines
1.7 KiB
C++
/*
|
|
* Copyright 2018 The WebRTC Project Authors. All rights reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include "api/crypto/cryptooptions.h"
|
|
#include "rtc_base/sslstreamadapter.h"
|
|
|
|
namespace webrtc {
|
|
|
|
CryptoOptions::CryptoOptions() {}
|
|
|
|
CryptoOptions::CryptoOptions(const CryptoOptions& other) {
|
|
enable_gcm_crypto_suites = other.enable_gcm_crypto_suites;
|
|
enable_encrypted_rtp_header_extensions =
|
|
other.enable_encrypted_rtp_header_extensions;
|
|
srtp = other.srtp;
|
|
}
|
|
|
|
CryptoOptions::~CryptoOptions() {}
|
|
|
|
// static
|
|
CryptoOptions CryptoOptions::NoGcm() {
|
|
CryptoOptions options;
|
|
options.srtp.enable_gcm_crypto_suites = false;
|
|
return options;
|
|
}
|
|
|
|
std::vector<int> CryptoOptions::GetSupportedDtlsSrtpCryptoSuites() const {
|
|
std::vector<int> crypto_suites;
|
|
if (srtp.enable_gcm_crypto_suites) {
|
|
crypto_suites.push_back(rtc::SRTP_AEAD_AES_256_GCM);
|
|
crypto_suites.push_back(rtc::SRTP_AEAD_AES_128_GCM);
|
|
}
|
|
// Note: SRTP_AES128_CM_SHA1_80 is what is required to be supported (by
|
|
// draft-ietf-rtcweb-security-arch), but SRTP_AES128_CM_SHA1_32 is allowed as
|
|
// well, and saves a few bytes per packet if it ends up selected.
|
|
// As the cipher suite is potentially insecure, it will only be used if
|
|
// enabled by both peers.
|
|
if (srtp.enable_aes128_sha1_32_crypto_cipher) {
|
|
crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_32);
|
|
}
|
|
crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_80);
|
|
return crypto_suites;
|
|
}
|
|
|
|
} // namespace webrtc
|