Index: webrtc/pc/srtptransport.h |
diff --git a/webrtc/pc/srtptransport.h b/webrtc/pc/srtptransport.h |
index 58ef205f872cc397474e65fd377403dae43dfaef..769df0620a00a37bdc51a5f5f400f18000ae1bc1 100644 |
--- a/webrtc/pc/srtptransport.h |
+++ b/webrtc/pc/srtptransport.h |
@@ -17,20 +17,17 @@ |
#include "webrtc/pc/rtptransportinternal.h" |
#include "webrtc/pc/srtpfilter.h" |
+#include "webrtc/pc/srtpsession.h" |
#include "webrtc/rtc_base/checks.h" |
namespace webrtc { |
// This class will eventually be a wrapper around RtpTransportInternal |
-// that protects and unprotects sent and received RTP packets. This |
-// functionality is currently implemented by SrtpFilter and BaseChannel, but |
-// will be moved here in the future. |
+// that protects and unprotects sent and received RTP packets. |
class SrtpTransport : public RtpTransportInternal { |
public: |
SrtpTransport(bool rtcp_mux_enabled, const std::string& content_name); |
- // TODO(zstein): Consider taking an RtpTransport instead of an |
- // RtpTransportInternal. |
SrtpTransport(std::unique_ptr<RtpTransportInternal> transport, |
const std::string& content_name); |
@@ -61,14 +58,21 @@ class SrtpTransport : public RtpTransportInternal { |
return rtp_transport_->GetRtcpPacketTransport(); |
} |
+ bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet, |
+ const rtc::PacketOptions& options, |
+ int flags) override; |
+ |
+ bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet, |
+ const rtc::PacketOptions& options, |
+ int flags) override; |
+ |
bool IsWritable(bool rtcp) const override { |
return rtp_transport_->IsWritable(rtcp); |
} |
- bool SendPacket(bool rtcp, |
- rtc::CopyOnWriteBuffer* packet, |
- const rtc::PacketOptions& options, |
- int flags) override; |
+ // The transport becomes active if the send_session_ and recv_session_ are |
+ // created. |
+ bool IsActive() const; |
bool HandlesPayloadType(int payload_type) const override { |
return rtp_transport_->HandlesPayloadType(payload_type); |
@@ -89,18 +93,104 @@ class SrtpTransport : public RtpTransportInternal { |
// TODO(zstein): Remove this when we remove RtpTransportAdapter. |
RtpTransportAdapter* GetInternal() override { return nullptr; } |
+ // Create new send/recv sessions and set the negotiated crypto keys for RTP |
+ // packet encryption. The keys can either come from SDES negotiation or DTLS |
+ // handshake. |
+ bool SetRtpParams(int send_cs, |
+ const uint8_t* send_key, |
+ int send_key_len, |
+ int recv_cs, |
+ const uint8_t* recv_key, |
+ int recv_key_len); |
+ |
+ // Create new send/recv sessions and set the negotiated crypto keys for RTCP |
+ // packet encryption. The keys can either come from SDES negotiation or DTLS |
+ // handshake. |
+ bool SetRtcpParams(int send_cs, |
+ const uint8_t* send_key, |
+ int send_key_len, |
+ int recv_cs, |
+ const uint8_t* recv_key, |
+ int recv_key_len); |
+ |
+ void ResetParams(); |
+ |
+ // Set the header extension ids that should be encrypted for the given source. |
+ // This method doesn't immediately update the SRTP session with the new IDs, |
+ // and you need to call SetRtpParams for that to happen. |
+ void SetEncryptedHeaderExtensionIds(cricket::ContentSource source, |
+ const std::vector<int>& extension_ids); |
+ |
+ // If external auth is enabled, SRTP will write a dummy auth tag that then |
+ // later must get replaced before the packet is sent out. Only supported for |
+ // non-GCM cipher suites and can be checked through "IsExternalAuthActive" |
+ // if it is actually used. This method is only valid before the RTP params |
+ // have been set. |
+ void EnableExternalAuth(); |
+ bool IsExternalAuthEnabled() const; |
+ |
+ // A SrtpTransport supports external creation of the auth tag if a non-GCM |
+ // cipher is used. This method is only valid after the RTP params have |
+ // been set. |
+ bool IsExternalAuthActive() const; |
+ |
+ // Returns srtp overhead for rtp packets. |
+ bool GetSrtpOverhead(int* srtp_overhead) const; |
+ |
+ // Returns rtp auth params from srtp context. |
+ bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len); |
+ |
+ // Helper method to get RTP Absoulute SendTime extension header id if |
+ // present in remote supported extensions list. |
+ void CacheRtpAbsSendTimeHeaderExtension(int rtp_abs_sendtime_extn_id) { |
+ rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id; |
+ } |
+ |
private: |
+ void CreateSrtpSessions(); |
+ |
void ConnectToRtpTransport(); |
+ bool SendPacket(bool rtcp, |
+ rtc::CopyOnWriteBuffer* packet, |
+ const rtc::PacketOptions& options, |
+ int flags); |
+ |
void OnPacketReceived(bool rtcp, |
rtc::CopyOnWriteBuffer* packet, |
const rtc::PacketTime& packet_time); |
void OnReadyToSend(bool ready) { SignalReadyToSend(ready); } |
- const std::string content_name_; |
+ bool ProtectRtp(void* data, int in_len, int max_len, int* out_len); |
+ |
+ // Overloaded version, outputs packet index. |
+ bool ProtectRtp(void* data, |
+ int in_len, |
+ int max_len, |
+ int* out_len, |
+ int64_t* index); |
+ bool ProtectRtcp(void* data, int in_len, int max_len, int* out_len); |
+ |
+ // Decrypts/verifies an invidiual RTP/RTCP packet. |
+ // If an HMAC is used, this will decrease the packet size. |
+ bool UnprotectRtp(void* data, int in_len, int* out_len); |
+ bool UnprotectRtcp(void* data, int in_len, int* out_len); |
+ |
+ const std::string content_name_; |
std::unique_ptr<RtpTransportInternal> rtp_transport_; |
+ |
+ std::unique_ptr<cricket::SrtpSession> send_session_; |
+ std::unique_ptr<cricket::SrtpSession> recv_session_; |
+ std::unique_ptr<cricket::SrtpSession> send_rtcp_session_; |
+ std::unique_ptr<cricket::SrtpSession> recv_rtcp_session_; |
+ |
+ std::vector<int> send_encrypted_header_extension_ids_; |
+ std::vector<int> recv_encrypted_header_extension_ids_; |
+ bool external_auth_enabled_ = false; |
+ |
+ int rtp_abs_sendtime_extn_id_ = -1; |
}; |
} // namespace webrtc |