Documentation
¶
Index ¶
- type MediaSessionWebrtc
- func (m *MediaSessionWebrtc) Close() error
- func (m *MediaSessionWebrtc) Codec() media.Codec
- func (m *MediaSessionWebrtc) CommonCodecs() []media.Codec
- func (m *MediaSessionWebrtc) Finalize(ctx context.Context) error
- func (m *MediaSessionWebrtc) Init(ctx context.Context, conf MediaSessionWebrtcConfig) error
- func (m *MediaSessionWebrtc) LocalSDP(_ context.Context, answered bool) ([]byte, error)
- func (m *MediaSessionWebrtc) ReadRTCP(buf []byte, pkts []rtcp.Packet) (int, error)
- func (m *MediaSessionWebrtc) ReadRTP(buf []byte, pkt *rtp.Packet) (int, error)
- func (m *MediaSessionWebrtc) ReadRTPRaw(buf []byte) (int, error)
- func (m *MediaSessionWebrtc) RemoteSDP(_ context.Context, body []byte, offered bool) error
- func (m *MediaSessionWebrtc) StartRTP(rw int8) error
- func (m *MediaSessionWebrtc) StopRTP(rw int8, duration time.Duration) error
- func (m *MediaSessionWebrtc) WriteRTCP(pkt rtcp.Packet) error
- func (m *MediaSessionWebrtc) WriteRTP(pkt *rtp.Packet) error
- type MediaSessionWebrtcConfig
- type RTPReadStats
- type RTPSessionWebrtc
- func (s *RTPSessionWebrtc) Close() error
- func (s *RTPSessionWebrtc) Monitor() error
- func (s *RTPSessionWebrtc) MonitorBackground() error
- func (s *RTPSessionWebrtc) MonitorClose() error
- func (s *RTPSessionWebrtc) OnReadRTCP(f func(pkt rtcp.Packet, rtpStats RTPReadStats))
- func (s *RTPSessionWebrtc) OnWriteRTCP(f func(pkt rtcp.Packet, rtpStats RTPWriteStats))
- func (s *RTPSessionWebrtc) ReadRTCP(buf []byte, pkts []rtcp.Packet) (int, error)
- func (s *RTPSessionWebrtc) ReadRTP(buf []byte, pkt *rtp.Packet) (int, error)
- func (s *RTPSessionWebrtc) ReadRTPRaw(buf []byte) (int, error)
- func (s *RTPSessionWebrtc) ReadStats() RTPReadStats
- func (s *RTPSessionWebrtc) WriteRTCP(pkt rtcp.Packet) error
- func (s *RTPSessionWebrtc) WriteRTP(pkt *rtp.Packet) error
- func (s *RTPSessionWebrtc) WriteStats() RTPWriteStats
- type RTPWriteStats
Constants ¶
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Variables ¶
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Functions ¶
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Types ¶
type MediaSessionWebrtc ¶
type MediaSessionWebrtc struct {
Laddr string
Raddr string
Codecs []media.Codec
Mode string
Config MediaSessionWebrtcConfig
// contains filtered or unexported fields
}
MediaSessionWebrtc is the direct ICE + DTLS-SRTP media stack. It intentionally does not embed MediaSession: legacy RTP uses two known UDP destinations, whereas WebRTC must send every DTLS/SRTP/SRTCP packet through the candidate pair selected and consent-checked by ICE.
Signalling lifecycle:
- Init gathers local candidates (non-trickle ICE).
- LocalSDP(false) creates an offer, or RemoteSDP(..., false) consumes one.
- RemoteSDP starts ICE as controlling for a local offer and controlled for a local answer. It returns before connectivity checks finish.
- Finalize waits for ICE, performs DTLS, verifies the SDP fingerprint and derives the SRTP keys. RTP must not be used before Finalize succeeds.
func (*MediaSessionWebrtc) Close ¶
func (m *MediaSessionWebrtc) Close() error
func (*MediaSessionWebrtc) Codec ¶
func (m *MediaSessionWebrtc) Codec() media.Codec
func (*MediaSessionWebrtc) CommonCodecs ¶
func (m *MediaSessionWebrtc) CommonCodecs() []media.Codec
func (*MediaSessionWebrtc) Finalize ¶
func (m *MediaSessionWebrtc) Finalize(ctx context.Context) error
Finalize completes ICE first and DTLS second. This order is the central WebRTC layering rule: ICE selects and maintains the network path; DTLS and all encrypted media then travel only over that path.
func (*MediaSessionWebrtc) Init ¶
func (m *MediaSessionWebrtc) Init(ctx context.Context, conf MediaSessionWebrtcConfig) error
Init creates the ICE agent and waits for candidate gathering to finish. SIP offer/answer has no standard trickle-ICE exchange, so the SDP returned later contains a complete candidate set and a=end-of-candidates.
func (*MediaSessionWebrtc) LocalSDP ¶
LocalSDP returns a complete, non-trickle WebRTC SDP. answered controls the DTLS setup attribute: an offer is actpass; an answer uses the role selected while parsing the remote offer.
func (*MediaSessionWebrtc) ReadRTPRaw ¶
func (m *MediaSessionWebrtc) ReadRTPRaw(buf []byte) (int, error)
func (*MediaSessionWebrtc) RemoteSDP ¶
RemoteSDP parses an answer when offered is true, otherwise it parses an offer. Starting ICE here is non-blocking so a SIP answer can be sent before the browser is required to complete connectivity checks.
func (*MediaSessionWebrtc) StartRTP ¶
func (m *MediaSessionWebrtc) StartRTP(rw int8) error
func (*MediaSessionWebrtc) StopRTP ¶
func (m *MediaSessionWebrtc) StopRTP(rw int8, duration time.Duration) error
type MediaSessionWebrtcConfig ¶
type MediaSessionWebrtcConfig struct {
ICEURLs []string
NetworkTypes []ice.NetworkType
PortMin uint16
PortMax uint16
IncludeLoopback bool
InterfaceFilter func(interfaceName string) bool
DTLS media.DTLSConfig
}
MediaSessionWebrtcConfig contains transport settings which do not belong to the codec-level media API. ICEURLs accepts STUN and TURN URLs understood by Pion (for example stun:stun.example.org:3478).
type RTPReadStats ¶
type RTPReadStats = media.RTPReadStats
RTPReadStats is shared with the core media package so existing statistics hooks can consume WebRTC and UDP RTP sessions uniformly.
type RTPSessionWebrtc ¶
type RTPSessionWebrtc struct {
// Keep pointers at the top to reduce GC scanning work.
Sess *MediaSessionWebrtc
// RTCPReportInterval controls sender/receiver report frequency. Set it
// before Monitor or MonitorBackground. Zero uses five seconds.
RTCPReportInterval time.Duration
// contains filtered or unexported fields
}
RTPSessionWebrtc adds RTP statistics and RTCP reporting to a negotiated MediaSessionWebrtc transport. It is separate from MediaSessionWebrtc so the ICE/DTLS/SRTP lifecycle does not own application-facing packet wrappers.
Unlike the legacy RTPSession this type has no symmetric-RTP address learning, RTP source lock or separate RTCP destination. ICE authenticates the selected path, while WebRTC sends both RTP and RTCP over that one path (rtcp-mux).
func NewRTPSessionWebrtc ¶
func NewRTPSessionWebrtc(sess *MediaSessionWebrtc) *RTPSessionWebrtc
func (*RTPSessionWebrtc) Close ¶
func (s *RTPSessionWebrtc) Close() error
Close requests RTCP monitoring to stop without waiting for the goroutines.
func (*RTPSessionWebrtc) Monitor ¶
func (s *RTPSessionWebrtc) Monitor() error
Monitor reads incoming SRTCP and periodically writes encrypted RTCP reports. It blocks until Close, a read failure or a write failure.
func (*RTPSessionWebrtc) MonitorBackground ¶
func (s *RTPSessionWebrtc) MonitorBackground() error
MonitorBackground starts the RTCP reader and report writer in the background. Call MonitorClose before closing MediaSessionWebrtc when deterministic cleanup is required.
func (*RTPSessionWebrtc) MonitorClose ¶
func (s *RTPSessionWebrtc) MonitorClose() error
MonitorClose stops RTCP monitoring and waits for both monitor goroutines. The ICE and media transports remain open.
func (*RTPSessionWebrtc) OnReadRTCP ¶
func (s *RTPSessionWebrtc) OnReadRTCP(f func(pkt rtcp.Packet, rtpStats RTPReadStats))
func (*RTPSessionWebrtc) OnWriteRTCP ¶
func (s *RTPSessionWebrtc) OnWriteRTCP(f func(pkt rtcp.Packet, rtpStats RTPWriteStats))
func (*RTPSessionWebrtc) ReadRTP ¶
ReadRTP decrypts through MediaSessionWebrtc and updates the receiver report counters. The generic RTPPacketReader can use RTPSessionWebrtc directly.
func (*RTPSessionWebrtc) ReadRTPRaw ¶
func (s *RTPSessionWebrtc) ReadRTPRaw(buf []byte) (int, error)
func (*RTPSessionWebrtc) ReadStats ¶
func (s *RTPSessionWebrtc) ReadStats() RTPReadStats
func (*RTPSessionWebrtc) WriteRTP ¶
func (s *RTPSessionWebrtc) WriteRTP(pkt *rtp.Packet) error
WriteRTP encrypts through MediaSessionWebrtc and updates sender-report counters only after the packet has been written successfully.
func (*RTPSessionWebrtc) WriteStats ¶
func (s *RTPSessionWebrtc) WriteStats() RTPWriteStats
type RTPWriteStats ¶
type RTPWriteStats = media.RTPWriteStats
RTPWriteStats is shared with the core media package so existing statistics hooks can consume WebRTC and UDP RTP sessions uniformly.
Source Files