Chromium Code Reviews| Index: webrtc/base/opensslstreamadapter.cc |
| diff --git a/webrtc/base/opensslstreamadapter.cc b/webrtc/base/opensslstreamadapter.cc |
| index 83efdc8bc0579493018eb7604ba99fb965e25d1a..4b226369f6177b0150ce34d9391cc47a6652f747 100644 |
| --- a/webrtc/base/opensslstreamadapter.cc |
| +++ b/webrtc/base/opensslstreamadapter.cc |
| @@ -523,6 +523,11 @@ void OpenSSLStreamAdapter::SetMaxProtocolVersion(SSLProtocolVersion version) { |
| ssl_max_version_ = version; |
| } |
| +void OpenSSLStreamAdapter::SetHandshakeTimeout(int timeout_ms) { |
| + RTC_DCHECK(ssl_ctx_ == NULL); |
|
Taylor Brandstetter
2017/02/02 18:07:41
Should leave a comment in the header file saying t
skvlad
2017/02/02 22:19:33
Done.
|
| + dtls_handshake_timeout_ms_ = timeout_ms; |
| +} |
| + |
| // |
| // StreamInterface Implementation |
| // |
| @@ -800,11 +805,10 @@ int OpenSSLStreamAdapter::BeginSSL() { |
| SSL_set_bio(ssl_, bio, bio); // the SSL object owns the bio now. |
| if (ssl_mode_ == SSL_MODE_DTLS) { |
| #ifdef OPENSSL_IS_BORINGSSL |
| - // Change the initial retransmission timer from 1 second to 50ms. |
| - // This will likely result in some spurious retransmissions, but |
| - // it's useful for ensuring a timely handshake when there's packet |
| - // loss. |
| - DTLSv1_set_initial_timeout_duration(ssl_, 50); |
| + // Change the initial retransmission timer to the value selected to match |
| + // the connection RTT. This ensures fast connection setup without excessive |
| + // bandwidth usage. |
|
Taylor Brandstetter
2017/02/02 18:07:41
This comment is a little out of place here; this c
skvlad
2017/02/02 22:19:33
Acknowledged.
|
| + DTLSv1_set_initial_timeout_duration(ssl_, dtls_handshake_timeout_ms_); |
| #else |
| // Enable read-ahead for DTLS so whole packets are read from internal BIO |
| // before parsing. This is done internally by BoringSSL for DTLS. |