Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(628)

Unified Diff: webrtc/base/sslstreamadapter.h

Issue 1337673002: Change WebRTC SslCipher to be exposed as number only. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Created 5 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/base/opensslstreamadapter.cc ('k') | webrtc/base/sslstreamadapter.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/base/sslstreamadapter.h
diff --git a/webrtc/base/sslstreamadapter.h b/webrtc/base/sslstreamadapter.h
index 4fb238a290482d13d00569bc57ccc5cdc45389f8..867f309a03b6c721339082e8a3676aa4772a8993 100644
--- a/webrtc/base/sslstreamadapter.h
+++ b/webrtc/base/sslstreamadapter.h
@@ -19,6 +19,23 @@
namespace rtc {
+// Constants for SRTP profiles.
+const uint16_t SRTP_AES128_CM_SHA1_80 = 0x0001;
+const uint16_t SRTP_AES128_CM_SHA1_32 = 0x0002;
+
+// Cipher suite to use for SRTP. Typically a 80-bit HMAC will be used, except
+// in applications (voice) where the additional bandwidth may be significant.
+// A 80-bit HMAC is always used for SRTCP.
+// 128-bit AES with 80-bit SHA-1 HMAC.
+extern const char CS_AES_CM_128_HMAC_SHA1_80[];
+// 128-bit AES with 32-bit SHA-1 HMAC.
+extern const char CS_AES_CM_128_HMAC_SHA1_32[];
+
+// Returns the DTLS-SRTP protection profile ID, as defined in
+// https://tools.ietf.org/html/rfc5764#section-4.1.2, for the given SRTP
+// Crypto-suite, as defined in https://tools.ietf.org/html/rfc4568#section-6.2
+uint16_t GetSrtpCryptoSuiteFromName(const std::string& cipher_rfc_name);
+
// SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS.
// After SSL has been started, the stream will only open on successful
// SSL verification of certificates, and the communication is
@@ -133,9 +150,9 @@ class SSLStreamAdapter : public StreamAdapterInterface {
// chain. The returned certificate is owned by the caller.
virtual bool GetPeerCertificate(SSLCertificate** cert) const = 0;
- // Retrieves the name of the cipher suite used for the connection
- // (e.g. "TLS_RSA_WITH_AES_128_CBC_SHA").
- virtual bool GetSslCipher(std::string* cipher);
+ // Retrieves the IANA registration id of the cipher suite used for the
+ // connection (e.g. 0x2F for "TLS_RSA_WITH_AES_128_CBC_SHA").
+ virtual bool GetSslCipherSuite(uint16_t* cipher);
// Key Exporter interface from RFC 5705
// Arguments are:
@@ -167,9 +184,14 @@ class SSLStreamAdapter : public StreamAdapterInterface {
// Returns the default Ssl cipher used between streams of this class
// for the given protocol version. This is used by the unit tests.
- // TODO(torbjorng@webrtc.org): Fix callers to avoid default parameter.
- static std::string GetDefaultSslCipher(SSLProtocolVersion version,
- KeyType key_type = KT_DEFAULT);
+ // TODO(guoweis): Move this away from a static class method.
+ static uint16_t GetDefaultSslCipherForTest(SSLProtocolVersion version,
+ KeyType key_type);
+
+ // TODO(guoweis): Move this away from a static class method. Currently this is
+ // introduced such that any caller could depend on sslstreamadapter.h without
+ // depending on specific SSL implementation.
+ static std::string GetSslCipherSuiteName(uint16_t cipher);
private:
// If true, the server certificate need not match the configured
« no previous file with comments | « webrtc/base/opensslstreamadapter.cc ('k') | webrtc/base/sslstreamadapter.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698