| Index: webrtc/base/sslstreamadapter.h
|
| diff --git a/webrtc/base/sslstreamadapter.h b/webrtc/base/sslstreamadapter.h
|
| index f6f0befa05a8d320da31255c06bf86a579be1584..9a9446267d34f9b1f3564d1c6c48a510fc1f745d 100644
|
| --- a/webrtc/base/sslstreamadapter.h
|
| +++ b/webrtc/base/sslstreamadapter.h
|
| @@ -24,8 +24,12 @@ const int TLS_NULL_WITH_NULL_NULL = 0;
|
|
|
| // Constants for SRTP profiles.
|
| const int SRTP_INVALID_CRYPTO_SUITE = 0;
|
| +#ifndef SRTP_AES128_CM_SHA1_80
|
| const int SRTP_AES128_CM_SHA1_80 = 0x0001;
|
| +#endif
|
| +#ifndef SRTP_AES128_CM_SHA1_32
|
| const int SRTP_AES128_CM_SHA1_32 = 0x0002;
|
| +#endif
|
|
|
| // 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.
|
|
|