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WebRTC: Support config, listener and SDP for TCP transport.

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This commit is contained in:
winlin 2022-09-02 22:39:19 +08:00
parent 424713a31d
commit 770d959148
14 changed files with 767 additions and 386 deletions
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74
trunk/src/app/srs_app_rtc_conn.hpp
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@ -52,6 +52,9 @@ class SrsRtcUserConfig;
class SrsRtcSendTrack;
class SrsRtcPublishStream;
class SrsEphemeralDelta;
class SrsRtcNetwork;
class SrsRtcUdpNetwork;
class ISrsRtcNetwork;
const uint8_t kSR = 200;
const uint8_t kRR = 201;
@ -101,12 +104,12 @@ public:
class SrsSecurityTransport : public ISrsRtcTransport
{
private:
SrsRtcConnection* session_;
ISrsRtcNetwork* network_;
SrsDtls* dtls_;
SrsSRTP* srtp_;
bool handshake_done;
public:
SrsSecurityTransport(SrsRtcConnection* s);
SrsSecurityTransport(ISrsRtcNetwork* s);
virtual ~SrsSecurityTransport();
srs_error_t initialize(SrsSessionConfig* cfg);
@ -136,7 +139,7 @@ private:
class SrsSemiSecurityTransport : public SrsSecurityTransport
{
public:
SrsSemiSecurityTransport(SrsRtcConnection* s);
SrsSemiSecurityTransport(ISrsRtcNetwork* s);
virtual ~SrsSemiSecurityTransport();
public:
srs_error_t protect_rtp(void* packet, int* nb_cipher);
@ -147,9 +150,9 @@ public:
class SrsPlaintextTransport : public ISrsRtcTransport
{
private:
SrsRtcConnection* session_;
ISrsRtcNetwork* network_;
public:
SrsPlaintextTransport(SrsRtcConnection* s);
SrsPlaintextTransport(ISrsRtcNetwork* s);
virtual ~SrsPlaintextTransport();
public:
virtual srs_error_t initialize(SrsSessionConfig* cfg);
@ -407,16 +410,6 @@ private:
void update_send_report_time(uint32_t ssrc, const SrsNtp& ntp, uint32_t rtp_time);
};
// Callback for RTC connection.
class ISrsRtcConnectionHijacker
{
public:
ISrsRtcConnectionHijacker();
virtual ~ISrsRtcConnectionHijacker();
public:
virtual srs_error_t on_dtls_done() = 0;
};
// A fast timer for conntion, for NACK feedback.
class SrsRtcConnectionNackTimer : public ISrsFastTimer
{
@ -444,11 +437,9 @@ private:
SrsRtcConnectionNackTimer* timer_nack_;
public:
bool disposing_;
ISrsRtcConnectionHijacker* hijacker_;
private:
SrsRtcServer* server_;
SrsRtcConnectionStateType state_;
ISrsRtcTransport* transport_;
private:
iovec* cache_iov_;
SrsBuffer* cache_buffer_;
@ -464,10 +455,8 @@ private:
private:
// The local:remote username, such as m5x0n128:jvOm where local name is m5x0n128.
std::string username_;
// The peer address, client maybe use more than one address, it's the current selected one.
SrsUdpMuxSocket* sendonly_skt;
// The address list, client may use multiple addresses.
std::map<std::string, SrsUdpMuxSocket*> peer_addresses_;
// Use one UDP network and one TCP network.
SrsRtcNetwork* network_;
private:
// TODO: FIXME: Rename it.
// The timeout of session, keep alive by STUN ping pong.
@ -485,14 +474,10 @@ private:
int twcc_id_;
// Simulators.
int nn_simulate_player_nack_drop;
// Pithy print for address change, use port as error code.
SrsErrorPithyPrint* pp_address_change;
// Pithy print for PLI request.
SrsErrorPithyPrint* pli_epp;
private:
bool nack_enabled_;
private:
SrsEphemeralDelta* delta_;
public:
SrsRtcConnection(SrsRtcServer* s, const SrsContextId& cid);
virtual ~SrsRtcConnection();
@ -511,8 +496,6 @@ public:
void set_state(SrsRtcConnectionStateType state);
// Get username pair for this connection, used as ID of session.
std::string username();
// Get all addresses client used.
std::vector<SrsUdpMuxSocket*> peer_addresses();
public:
virtual ISrsKbpsDelta* delta();
// Interface ISrsResource.
@ -532,7 +515,7 @@ public:
// Before initialize, user must set the local SDP, which is used to inititlize DTLS.
srs_error_t initialize(SrsRequest* r, bool dtls, bool srtp, std::string username);
// The peer address may change, we can identify that by STUN messages.
srs_error_t on_stun(SrsUdpMuxSocket* skt, SrsStunPacket* r);
srs_error_t on_stun(SrsStunPacket* r, char* data, int nb_data);
srs_error_t on_dtls(char* data, int nb_data);
srs_error_t on_rtp(char* data, int nb_data);
private:
@ -545,16 +528,13 @@ private:
public:
srs_error_t on_rtcp_feedback_twcc(char* buf, int nb_buf);
srs_error_t on_rtcp_feedback_remb(SrsRtcpPsfbCommon *rtcp);
public:
void set_hijacker(ISrsRtcConnectionHijacker* h);
public:
srs_error_t on_connection_established();
srs_error_t on_dtls_alert(std::string type, std::string desc);
srs_error_t start_play(std::string stream_uri);
srs_error_t start_publish(std::string stream_uri);
bool is_alive();
void alive();
void update_sendonly_socket(SrsUdpMuxSocket* skt);
public:
SrsRtcUdpNetwork* udp();
public:
// send rtcp
srs_error_t send_rtcp(char *data, int nb_data);
@ -582,33 +562,5 @@ private:
srs_error_t create_publisher(SrsRequest* request, SrsRtcSourceDescription* stream_desc);
};
class ISrsRtcHijacker
{
public:
ISrsRtcHijacker();
virtual ~ISrsRtcHijacker();
public:
// Initialize the hijacker.
virtual srs_error_t initialize() = 0;
// When create publisher, SDP is done, DTLS is not ready.
virtual srs_error_t on_create_publish(SrsRtcConnection* session, SrsRtcPublishStream* publisher, SrsRequest* req) = 0;
// When start publisher by RTC, SDP and DTLS are done.
virtual srs_error_t on_start_publish(SrsRtcConnection* session, SrsRtcPublishStream* publisher, SrsRequest* req) = 0;
// When stop publish by RTC.
virtual void on_stop_publish(SrsRtcConnection* session, SrsRtcPublishStream* publisher, SrsRequest* req) = 0;
// When got RTP plaintext packet.
virtual srs_error_t on_rtp_packet(SrsRtcConnection* session, SrsRtcPublishStream* publisher, SrsRequest* req, SrsRtpPacket* pkt) = 0;
// When before play by RTC. (wait source to ready in cascade scenario)
virtual srs_error_t on_before_play(SrsRtcConnection* session, SrsRequest* req) = 0;
// When start player by RTC.
virtual srs_error_t on_start_play(SrsRtcConnection* session, SrsRtcPlayStream* player, SrsRequest* req) = 0;
// When stop player by RTC.
virtual void on_stop_play(SrsRtcConnection* session, SrsRtcPlayStream* player, SrsRequest* req) = 0;
// When start consuming for player for RTC.
virtual srs_error_t on_start_consume(SrsRtcConnection* session, SrsRtcPlayStream* player, SrsRequest* req, SrsRtcConsumer* consumer) = 0;
};
extern ISrsRtcHijacker* _srs_rtc_hijacker;
#endif