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WebRTC: Extract SrsRtcNetwork layer for UDP/TCP.

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This commit is contained in:
winlin 2022-09-03 11:52:11 +08:00
parent 770d959148
commit 625069af7f
23 changed files with 698 additions and 528 deletions
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39
trunk/src/app/srs_app_rtc_conn.hpp
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@ -52,7 +52,7 @@ class SrsRtcUserConfig;
class SrsRtcSendTrack;
class SrsRtcPublishStream;
class SrsEphemeralDelta;
class SrsRtcNetwork;
class SrsRtcNetworks;
class SrsRtcUdpNetwork;
class ISrsRtcNetwork;
@ -67,17 +67,6 @@ const uint8_t kRtpFb = 205;
const uint8_t kPsFb = 206;
const uint8_t kXR = 207;
enum SrsRtcConnectionStateType
{
// TODO: FIXME: Should prefixed by enum name.
INIT = -1,
WAITING_ANSWER = 1,
WAITING_STUN = 2,
DOING_DTLS_HANDSHAKE = 3,
ESTABLISHED = 4,
CLOSED = 5,
};
// The transport for RTC connection.
class ISrsRtcTransport : public ISrsDtlsCallback
{
@ -378,10 +367,8 @@ private:
srs_error_t send_rtcp_rr();
srs_error_t send_rtcp_xr_rrtr();
public:
srs_error_t on_rtp(char* buf, int nb_buf);
private:
// @remark We copy the plaintext, user should free it.
srs_error_t on_rtp_plaintext(char* plaintext, int nb_plaintext);
srs_error_t on_rtp_cipher(char* buf, int nb_buf);
srs_error_t on_rtp_plaintext(char* buf, int nb_buf);
private:
srs_error_t do_on_rtp_plaintext(SrsRtpPacket*& pkt, SrsBuffer* buf);
public:
@ -439,7 +426,6 @@ public:
bool disposing_;
private:
SrsRtcServer* server_;
SrsRtcConnectionStateType state_;
private:
iovec* cache_iov_;
SrsBuffer* cache_buffer_;
@ -455,8 +441,8 @@ private:
private:
// The local:remote username, such as m5x0n128:jvOm where local name is m5x0n128.
std::string username_;
// Use one UDP network and one TCP network.
SrsRtcNetwork* network_;
// A group of networks, each has its own DTLS and SRTP context.
SrsRtcNetworks* networks_;
private:
// TODO: FIXME: Rename it.
// The timeout of session, keep alive by STUN ping pong.
@ -491,9 +477,8 @@ public:
void set_local_sdp(const SrsSdp& sdp);
SrsSdp* get_remote_sdp();
void set_remote_sdp(const SrsSdp& sdp);
// Connection level state machine, for ARQ of UDP packets.
SrsRtcConnectionStateType state();
void set_state(SrsRtcConnectionStateType state);
// Change network to waiting stun state.
void set_state_as_waiting_stun();
// Get username pair for this connection, used as ID of session.
std::string username();
public:
@ -514,10 +499,8 @@ public:
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(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);
srs_error_t on_rtp_cipher(char* data, int nb_data);
srs_error_t on_rtp_plaintext(char* data, int nb_data);
private:
// Decode the RTP header from buf, find the publisher by SSRC.
srs_error_t find_publisher(char* buf, int size, SrsRtcPublishStream** ppublisher);
@ -549,8 +532,10 @@ public:
srs_error_t do_send_packet(SrsRtpPacket* pkt);
// Directly set the status of play track, generally for init to set the default value.
void set_all_tracks_status(std::string stream_uri, bool is_publish, bool status);
public:
// Notify by specified network.
srs_error_t on_binding_request(SrsStunPacket* r, std::string& ice_pwd);
private:
srs_error_t on_binding_request(SrsStunPacket* r);
// publish media capabilitiy negotiate
srs_error_t negotiate_publish_capability(SrsRtcUserConfig* ruc, SrsRtcSourceDescription* stream_desc);
srs_error_t generate_publish_local_sdp(SrsRequest* req, SrsSdp& local_sdp, SrsRtcSourceDescription* stream_desc, bool unified_plan);