Index: pc/srtptransport.h |
diff --git a/pc/srtptransport.h b/pc/srtptransport.h |
index 9f20e1d8f662cd5cfc20139f5f88e99360be54f9..4d2ecb813b917e887499ccabff5dc8b7ebd200a8 100644 |
--- a/pc/srtptransport.h |
+++ b/pc/srtptransport.h |
@@ -17,17 +17,20 @@ |
#include "pc/rtptransportinternal.h" |
#include "pc/srtpfilter.h" |
-#include "pc/srtpsession.h" |
#include "rtc_base/checks.h" |
namespace webrtc { |
// This class will eventually be a wrapper around RtpTransportInternal |
-// that protects and unprotects sent and received RTP packets. |
+// 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. |
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); |
@@ -58,21 +61,14 @@ 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); |
} |
- // The transport becomes active if the send_session_ and recv_session_ are |
- // created. |
- bool IsActive() const; |
+ bool SendPacket(bool rtcp, |
+ rtc::CopyOnWriteBuffer* packet, |
+ const rtc::PacketOptions& options, |
+ int flags) override; |
bool HandlesPayloadType(int payload_type) const override { |
return rtp_transport_->HandlesPayloadType(payload_type); |
@@ -93,104 +89,18 @@ 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); } |
- 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; |
+ std::unique_ptr<RtpTransportInternal> rtp_transport_; |
}; |
} // namespace webrtc |