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Unified Diff: webrtc/pc/srtptransport.h

Issue 2997983002: Completed the functionalities of SrtpTransport. (Closed)
Patch Set: Fix the chromimum issue. Created 3 years, 4 months ago
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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
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