mrtc — RFC Compliance Report

Status of each IETF RFC relevant to the WebRTC protocol stack as implemented in mrtc.

Legend:
✅ Implemented · ⚠️ Partial · ❌ Not Implemented · 📋 Planned

Summary

RFC	Title	Status	Coverage	Successor
RFC 3550	RTP / RTCP	⚠️ Partial	~50%	Updated by 10 RFCs
RFC 3711	SRTP	⚠️ Partial	~75%	Updated by 6904, 8723
RFC 4566	SDP	⚠️ Partial	~70%	Obsoleted by 8866
RFC 4585	RTCP Feedback (NACK, PLI, FIR)	⚠️ Partial	NACK/PLI/FIR implemented	Updated by 5506, 8108
RFC 5109	UlpFEC	❌ Not Impl	0%	Current
RFC 5389	STUN	✅ Implemented	~95%	Obsoleted by 8489
RFC 5764	DTLS-SRTP	✅ Implemented	~95%	Updated by 7983
RFC 5766	TURN	⚠️ Partial	~50%	Obsoleted by 8656
RFC 6347	DTLS 1.2	✅ Implemented	~95%	Obsoleted by 9147
RFC 6455	WebSocket (signaling)	✅ Implemented	via libwebsockets	Updated by 8441
RFC 7587	Opus RTP Payload	✅ Implemented	~90%	Current
RFC 7714	SRTP AES-GCM	✅ Implemented	~90%	Current
RFC 7741	VP8 RTP Payload	⚠️ Partial	~85%	Current
RFC 7983	Multiplexing STUN/DTLS/RTP	✅ Implemented	100%	Current
RFC 8445	ICE	⚠️ Partial	~45%	Updated by 8863
RFC 8285	One-byte/two-byte RTP header extensions	⚠️ Partial	one-byte form	Current
RFC 8829	JSEP (offer/answer model)	⚠️ Partial	~60%	Current
RFC 8834	Media Transport over RTP	⚠️ Partial	~50%	Current
RFC 8839	ICE SDP Attributes	⚠️ Partial	~60%	Current
draft-holmer-rmcat-transport-wide-cc	Transport-Wide Congestion Control	⚠️ Partial	wire + GCC	(was RFC 8888 row)

RFC 3550 — RTP/RTCP

"RTP: A Transport Protocol for Real-Time Applications"

Updated by: RFC 5506 (reduced-size RTCP), RFC 5761 (RTP/RTCP mux on single port), RFC 6051 (rapid RTP sync), RFC 6222/7022 (CNAME generation), RFC 7160 (multiple media types), RFC 7164 (RTP/RTCP mux considerations), RFC 8083 (multimedia congestion feedback), RFC 8108 (SSRC-specific RTCP), RFC 8860 (deprecates RTP/RTCP transport address pairing). Still the base specification — not obsoleted.

Files: rtc/src/rtc_rtp.c, rtc/src/rtc_rtcp.c, rtc/include/rtc/rtc_rtp.h, rtc/include/rtc/rtc_rtcp.h

Implemented

Section	Feature	Notes
5.1	RTP fixed header (V, P, X, CC, M, PT, seq, ts, SSRC)	Full 12-byte header serialization and parsing
5.1	Version field = 2	Enforced on build and validated on parse
5.1	Marker bit	Set/get supported, used for frame boundaries
5.1	Payload type	Dynamic PT via `rtc_codec_t`
5.1	Sequence number auto-increment	Per-session tracking in `rtc_rtp_session_t`
5.1	Timestamp auto-increment	Per-codec clock rate (e.g. 48000 for Opus, 90000 for VP8)
5.1	SSRC random generation	Per-session unique SSRC
6.4.1	Sender Report (SR, PT=200)	Build and parse with NTP timestamp, RTP timestamp, packet/octet counts
6.4.2	Receiver Report (RR, PT=201)	Build and parse with fraction lost, cumulative lost, highest seq, jitter, LSR, DLSR
A.8	Interarrival jitter calculation	Per-packet jitter estimate per RFC formula
—	Periodic RTCP transmission	Every 5 seconds from peer connection
—	RTCP multiplexed with RTP	Single UDP socket via `a=rtcp-mux`

Partially Implemented

Section	Feature	Status
5.1	CSRC list (Contributing Sources)	CC field parsed but CSRC entries ignored
5.3	RTP header extensions (X bit)	Extension bit detected but extension data not parsed
5.1	Padding (P bit)	P bit handled in header but padding bytes not stripped
6.4.1	SR NTP timestamp accuracy	Approximated from monotonic clock, not wall-clock
6.4.2	RR loss calculation	Assumes seq starts at 0; `packets_expected = highest_seq + 1` is wrong if initial seq ≠ 0
6.1	Compound RTCP packets	Only single SR or RR sent per interval, not compound packets

Not Implemented

Section	Feature	Status
6.1	Compound RTCP requirement	Spec requires SR/RR + SDES in every compound packet
6.5	SDES (Source Description, PT=202)	Constant defined (`RTCP_PT_SDES = 202`) but no build/parse
6.6	BYE (PT=203)	Not implemented
6.7	APP (Application-Defined, PT=204)	Not implemented
6.2	RTCP transmission interval calculation	Fixed 5-second interval; spec requires bandwidth-adaptive timing
6.3.5	RTCP timer reconsideration	Not implemented
6.2.1	Minimum RTCP interval (5s reduced by random factor)	Not randomized
7	Translators and mixers	Not applicable (endpoint-only)
8	SSRC collision detection	Not implemented
13	RTP over TCP	Not implemented (UDP only)

RFC 3711 — SRTP

"The Secure Real-time Transport Protocol"

Updated by: RFC 6904 (encrypted SRTP header extensions), RFC 8723 (SRTP double encryption). Still the base specification — not obsoleted.

Files: rtc/src/rtc_srtp.c, rtc/src/rtc_srtp.h

Implemented

Section	Feature	Notes
3.1	SRTP packet format (RTP header + encrypted payload + auth tag)	Full protection/unprotection
3.3	Authentication tag (HMAC-SHA1-80, 10 bytes)	Appended on protect, verified on unprotect
3.4	SRTCP (encrypted RTCP + SRTCP index + auth tag)	Full protect/unprotect with E-flag
4.1.1	IV construction (salt ⊕ (SSRC ∥ packet_index))	Correctly implemented for AES-128-CM
4.1.1	AES-128-CM (Counter Mode) encryption	Using OpenSSL EVP API
—	AEAD_AES_128_GCM cipher (RFC 7714)	Preferred profile; see RFC 7714 section
4.3.1	Key Derivation Function (KDF)	AES-CM PRF with label-based derivation
4.3.1	Session key, salt, and auth key derivation	Separate derivation for RTP and RTCP
3.3.1	Rollover Counter (ROC) for 48-bit index	ROC tracking on 16-bit sequence wrap
—	Session key caching	Keys cached and re-derived only when necessary

Not Implemented

Section	Feature	Impact
3.3.2	Replay protection (replay list)	CRITICAL — Captured packets can be replayed. RFC requires a replay list with configurable window (default 64).
3.3.1	ROC guessing for out-of-order packets	Partial — only simple wrap detection, no extended guessing algorithm
4.1.2	AES-256-CM cipher suite	Only AES-128-CM supported
4.1.3	NULL cipher (for debugging)	Not supported
4.2.1	HMAC-SHA1-32 (truncated to 32 bits)	Only 80-bit tag supported
9	Master Key Index (MKI)	Not implemented; single master key assumed
9.1	Key management / rekeying	No key rotation mechanism
3.1	SRTP buffer size validation	`protect` appends auth tag without verifying buffer capacity

RFC 7714 — SRTP AES-GCM

"AES-GCM Authenticated Encryption in the Secure Real-time Transport Protocol (SRTP)"

Status: ✅ Implemented (~90%)

Files: rtc/src/rtc_srtp.c, rtc/src/rtc_srtp.h, rtc/src/rtc_dtls.c

Implemented

Section	Feature	Notes
8.1	SRTP AEAD_AES_128_GCM / AEAD_AES_256_GCM (12-byte IV, 16-byte tag)	RTP header as associated data; payload encrypted in place
9.1	SRTCP AEAD_AES_*_GCM	Header + E\|index as associated data; E-flag honored
11	16/32-byte key, 12-byte salt key derivation	Same SRTP KDF (AES-128 or AES-256 PRF by key size), no separate auth key
—	DTLS-SRTP profile negotiation	`SRTP_AEAD_AES_256_GCM` then `SRTP_AEAD_AES_128_GCM` offered first (preferred), selected per handshake

Not Implemented

Feature	Notes
AEAD with 192-bit keys	Only AES-128 / AES-256 GCM are offered

RFC 4566 — SDP

"SDP: Session Description Protocol"

⚠️ Obsoleted by RFC 8866 (January 2021). RFC 8866 is a clarification update with no wire-format changes. Key differences: removes k= line entirely, clarifies b= semantics, tightens ABNF grammar, mandates ICE-era usage patterns. Migration effort: low — mrtc's minimal SDP subset is unaffected; simply update doc references.

Files: rtc/src/rtc_sdp.c, rtc/include/rtc/rtc_sdp.h

Implemented

Section	Feature	Notes
5.1	`v=` (protocol version = 0)	Generated and parsed
5.2	`o=` (origin: username, session ID, version, network type, address)	Generated with fixed values
5.3	`s=` (session name)	Fixed "mrtc"
5.7	`t=` (timing = "0 0")	Generated
5.14	`m=` (media descriptions: audio, video, application)	Multi-media line generation and parsing
5.7	`c=` (connection data)	Placeholder `0.0.0.0`
5.13	`a=` (attributes)	Extensive attribute support (see below)
—	`a=rtpmap:` codec parameters	Payload type, codec name, clock rate, channels
—	`a=fingerprint:` DTLS fingerprint	SHA-256 hex fingerprint
—	`a=setup:` DTLS role	active / passive / actpass
—	`a=ice-ufrag:` / `a=ice-pwd:`	Per-media ICE credentials
—	`a=candidate:` lines	Full candidate formatting and parsing
—	`a=mid:` media identification	Per-media section MID tag
—	`a=group:BUNDLE`	BUNDLE group for all media sections
—	`a=rtcp-mux`	Generated for all media sections
—	`a=sendrecv` / `a=sendonly` / `a=recvonly` / `a=inactive`	Direction attribute generation
—	`m=application ... webrtc-datachannel`	Data channel media line with `a=sctp-port:5000`

Not Implemented

Section	Feature	Notes
5.8	`b=` (bandwidth)	Not generated or parsed
5.2	Origin version incrementing	Fixed version, not incremented on reoffer
5.4	`i=` (session/media information)	Not generated
5.5	`u=` (URI)	Not generated
5.6	`e=` / `p=` (email/phone)	Not generated
5.9	`z=` (time zones)	Not generated
5.10	`k=` (encryption keys) — deprecated	Not generated (correct to omit)
—	`a=fmtp:` (format parameters)	Not generated; no codec-specific params
—	`a=extmap:` (RTP header extensions)	Generated and parsed for negotiated RTP extensions
—	`a=ssrc:` (SSRC attributes)	Generated and parsed for audio/video media
—	`a=rtcp-fb:` (RTCP feedback capabilities)	Not generated; needed for NACK/PLI/FIR
—	`a=ice-options:trickle`	Generated and parsed
—	Full SDP parsing	Minimal parser sufficient for WebRTC; not RFC-complete

Known Limitations

SDP parsing remains intentionally minimal and accepts only the subset needed by this stack.
Relay candidates labeled "host": Missing ICE_CANDIDATE_RELAY case defaults to "host" type string.

RFC 4585 — RTCP Feedback

"Extended RTP Profile for RTCP-Based Feedback (RTP/AVPF)"

Updated by: RFC 5506 (allows reduced-size RTCP-FB without compound), RFC 8108 (SSRC-specific attributes). Still the base specification — not obsoleted.

Feature	Status
Generic NACK (negative acknowledgement)	✅ Build, parse, dispatch implemented
PLI (Picture Loss Indication)	✅ Build, parse, dispatch implemented
FIR (Full Intra Request)	✅ Build, parse, dispatch implemented
NACK retransmit buffer	✅ Per-video sender buffer with logical raw retransmit
`a=rtcp-fb:` SDP negotiation	❌ Not implemented

RFC 5109 — UlpFEC

"RTP Payload Format for Generic Forward Error Correction"

📋 Planned for Phase 4 (see ROADMAP.md)

Feature	Status
XOR-based FEC packet generation	❌ Not implemented
FEC recovery at receiver	❌ Not implemented
`a=rtpmap:XX ulpfec/90000` SDP	❌ Not implemented

RFC 5389 — STUN

"Session Traversal Utilities for NAT"

⚠️ Obsoleted by RFC 8489 (February 2020). Key changes: long-term auth mechanism v2 (SHA-256 + PASSWORD-ALGORITHMS/PASSWORD-ALGORITHM attributes), ALTERNATE-DOMAIN attribute, STUN test vectors updated, nonce rotation required. Migration effort: medium — current long-term auth (MD5) still works but new servers may require auth v2.

Files: rtc/src/rtc_stun.c, rtc/src/rtc_stun.h

Implemented

Section	Feature	Notes
6	STUN message format (20-byte header: type, length, magic cookie, transaction ID)	Full serialization and parsing
6	Magic Cookie (`0x2112A442`)	Validated on parse
6.1	Message types: Binding Request (0x0001), Success Response (0x0101)	Build and parse
15.1	MAPPED-ADDRESS (0x0001)	Parsed
15.2	XOR-MAPPED-ADDRESS (0x0020)	Build and parse with XOR encoding
15.4	MESSAGE-INTEGRITY (0x0008)	HMAC-SHA1 generation and verification
15.5	FINGERPRINT (0x8028)	CRC-32 ⊕ 0x5354554E generation and verification
15.3	USERNAME (0x0006)	Build for ICE connectivity checks
15.6	ERROR-CODE (0x0009)	Parsed (used for TURN 401 handling)
15.10	SOFTWARE (0x8022)	Written as "mrtc"
—	STUN Binding Request to public server	Blocking request with timeout
—	Transaction ID generation	Random 96-bit IDs
—	Long-term credential mechanism (MD5(user:realm:pass))	For TURN authentication

ICE-Specific Attributes (RFC 8445)

Attribute	Code	Status
PRIORITY (0x0024)	Implemented	Written in connectivity check requests
USE-CANDIDATE (0x0025)	Implemented	Flag attribute for nomination
ICE-CONTROLLED (0x8029)	Implemented	With tie-breaker
ICE-CONTROLLING (0x802A)	Implemented	With tie-breaker (always used)

TURN Attributes (RFC 5766)

Attribute	Code	Status
CHANNEL-NUMBER (0x000C)	✅
LIFETIME (0x000D)	✅
XOR-PEER-ADDRESS (0x0012)	✅
DATA (0x0013)	✅
REALM (0x0014)	✅
NONCE (0x0015)	✅
XOR-RELAYED-ADDRESS (0x0016)	✅
REQUESTED-TRANSPORT (0x0019)	✅

Not Implemented

Section	Feature	Notes
7.2.2	Error response handling (beyond 401)	Only 401 Unauthorized handled for TURN
7.2.1	Retransmission (RTO doubling)	Single attempt with fixed timeout
10	Short-term credential mechanism	Not used (long-term credentials only)
15.7	REALM attribute build	Parsed from server response but not self-generated
15.8	NONCE attribute build	Parsed from server response but not self-generated
—	STUN over TCP/TLS	UDP only
—	ALTERNATE-SERVER (0x8023)	Not handled

Known Bug

IPv6 OOB read: Checks alen >= 8 but IPv6 MAPPED-ADDRESS needs 20 bytes. Reads past buffer for IPv6 addresses.

RFC 5764 — DTLS-SRTP

"Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)"

Updated by: RFC 7983 (demux clarification — already implemented by mrtc). Still the base specification — not obsoleted.

Files: rtc/src/rtc_dtls.c, rtc/src/rtc_dtls.h

Implemented

Section	Feature	Notes
4.1	SRTP protection profile negotiation	`SRTP_AEAD_AES_256_GCM` / `SRTP_AEAD_AES_128_GCM` (preferred) + `SRTP_AES128_CM_SHA1_80` via `SSL_CTX_set_tlsext_use_srtp`; selected profile read back with `SSL_get_selected_srtp_profile`
4.2	Key material export (`EXTRACTOR-dtls_srtp`)	`SSL_export_keying_material` called after handshake
4.2	Client/server key material split	Correct split into client_write/server_write key+salt pairs
—	SDP `a=fingerprint:sha-256`	SHA-256 fingerprint extracted and exchanged
—	SDP `a=setup:` role signaling	active/passive/actpass role from SDP

Not Implemented

Feature	Notes
Fingerprint verification against SDP	Callback always accepts; fingerprint compared separately (if at all)

RFC 5766 — TURN

"Traversal Using Relays around NAT"

⚠️ Obsoleted by RFC 8656 (February 2020). Key changes: TURN over TCP/TLS/DTLS transports, dual-stack (IPv4+IPv6) allocation, REQUESTED-ADDRESS-FAMILY attribute, uses RFC 8489 auth. Migration effort: medium — current UDP-only TURN client works but modern TURN servers may require RFC 8489 long-term auth v2 and offer TURN-over-TLS.

Files: rtc/src/rtc_turn.c, rtc/src/rtc_turn.h

Implemented (Client)

Section	Feature	Notes
6	Allocate Request	With 401 challenge-response for external TURN services
7	Refresh Request	Including lifetime=0 for deallocation
9	CreatePermission Request	Single peer address
11	ChannelBind Request	Channel number 0x4000
11.4	ChannelData framing	4-byte header (channel + length) + payload
—	Long-term credential mechanism	MD5(username:realm:password)
—	401 Unauthorized challenge response	Realm + nonce extraction and retry

Not Implemented

Section	Feature	Impact
5	Relay candidate integration with ICE	TURN allocations are not wired into ICE candidate gathering
7.2	Permission refresh timers (5 min)	No timer-based refresh — permissions expire silently
11.3	Channel refresh timers (10 min)	No timer-based refresh — channels expire
7.1	Allocation refresh timers	No automatic refresh before allocation expires
—	Multiple allocations / permissions	Single allocation per client, single permission
—	Event-driven integration	Blocking socket operations, not integrated with transport I/O poller
—	TCP relaying	UDP only
—	TURN over TLS/DTLS	UDP only

Known Bugs

Auth key truncation: Binary MD5 lt_key passed as const char* to HMAC functions using strlen(). Null bytes truncate key (~63% chance of breakage).
Nonce never rotated (server): Single nonce for server lifetime enables replay attacks.

RFC 6347 — DTLS 1.2

"Datagram Transport Layer Security Version 1.2"

⚠️ Obsoleted by RFC 9147 (DTLS 1.3, April 2022). DTLS 1.3 aligns with TLS 1.3: 0-RTT, encrypted handshake, removed static RSA/DH, unified record layer. Migration effort: high — requires OpenSSL 3.2+ with DTLS 1.3 support (not yet widely available). DTLS 1.2 remains valid and interoperable for WebRTC; browsers still support it.

Files: rtc/src/rtc_dtls.c, rtc/src/rtc_dtls.h

Implemented

Feature	Notes
DTLS handshake (client and server roles)	Full handshake via OpenSSL memory BIOs
Self-signed EC certificate (P-256)	Modern `EVP_PKEY_CTX` API (OpenSSL 3.0+)
SHA-256 fingerprint extraction	Hex-formatted for SDP exchange
Flight retransmission	`DTLSv1_handle_timeout` + timer integration
State machine	NEW → CONNECTING → CONNECTED / FAILED / CLOSED
Memory BIO mode	Application controls UDP transmission
SRTP profile registration	`SRTP_AES128_CM_SHA1_80` via `SSL_CTX_set_tlsext_use_srtp`
Application data after handshake	`SSL_read` / `SSL_write` for data channel messages

Not Implemented

Feature	Notes
DTLS 1.3	Uses DTLS 1.2 only
Client certificate verification	Callback always accepts (`SSL_VERIFY_PEER` with dummy callback)
Connection ID extension	Not used
DTLS renegotiation	Not supported

RFC 6455 — WebSocket

"The WebSocket Protocol"

Updated by: RFC 7936 (clarification), RFC 8307 (well-known URIs), RFC 8441 (WebSocket bootstrapping over HTTP/2). Still the base specification. Migration effort: none — handled by libwebsockets.

Files: signaling/src/signaling_client.c, signaling/server/signaling_server.c

Implemented

Feature	Notes
WebSocket client connections	Via libwebsockets library
WebSocket server	Via libwebsockets library
Text message framing	JSON signaling messages
Connection lifecycle	Open, message, close callbacks

Implementation delegated entirely to libwebsockets. No raw RFC 6455 framing code.

RFC 7587 — Opus RTP Payload

"RTP Payload Format for the Opus Speech and Audio Codec"

Files: media/src/codec_opus.c, media/src/audio_codec.c

Implemented

Feature	Notes
Opus encoding/decoding	libopus, 48 kHz fixed sample rate
RTP packetization	One Opus frame per RTP packet
Mono and stereo	Configurable channels
Variable bitrate	Opus VBR mode
20ms frame duration	Fixed at 960 samples per frame (48kHz × 20ms)

Not Implemented

Feature	Notes
`a=fmtp:` Opus parameters	No `maxaveragebitrate`, `stereo`, `useinbandfec` signaling
FEC (in-band Opus FEC)	Opus supports built-in FEC; not enabled
DTX (Discontinuous Transmission)	Not enabled
CBR mode	Always VBR
Multiple frames per packet	Single frame only

RFC 7741 — VP8 RTP Payload

"RTP Payload Format for VP8 Video"

Files: media/src/vp8_packetizer.c, media/src/vp8_packetizer.h, media/src/codec_vp8.c

Implemented

Section	Feature	Notes
4.2	VP8 payload descriptor (1-byte)	S-bit (start of partition), PID
4.2	Marker bit (M) on last packet of frame	Correct frame boundary signaling
—	Packetization (split frame into MTU-sized RTP packets)	Max 1200 bytes per packet
—	Depacketization (reassemble frame from RTP packets)	Sequence-ordered reassembly
—	Keyframe detection	Via VP8 payload header (P-bit in first byte)
—	VP8 encoding/decoding	libvpx integration

Not Implemented

Section	Feature	Impact
4.2	2-byte PictureID (M-bit extension)	Breaks interop with peers sending extended PictureID
4.2	TL0PICIDX (temporal layer index)	No temporal scalability
4.2	TID (temporal layer ID)	No SVC support
4.2	KEYIDX (key frame index)	Not parsed
4.4	VP8 payload descriptor extensions (X-bit)	I, L, T, K extension bits ignored

RFC 7983 — Multiplexing STUN/DTLS/RTP/TURN

"Multiplexing Scheme Updates for Secure Real-time Transport Protocol (SRTP) Extension for Datagram Transport Layer Security (DTLS)"

Files: rtc/src/rtc_transport.c, rtc/src/rtc_listener.c

Implemented

Feature	Notes
First-byte classification	Full demux table: STUN (0–3), DTLS (20–63), TURN ChannelData (64–79), RTP/RTCP (128–191)
Single UDP socket multiplexing	Shared listeners demux many logical transports on one UDP socket
Callback dispatch per packet type	STUN routed by ufrag/transaction, DTLS/RTP/RTCP routed by selected tuple

Fully compliant with RFC 7983 packet classification.

RFC 8445 — ICE

"Interactive Connectivity Establishment"

Updated by: RFC 8863 (Trickle ICE, January 2021). Defines incremental candidate exchange during gathering. Migration effort: medium — requires implementing add_ice_candidate(), SDP a=ice-options:trickle, and a=end-of-candidates signaling.

Files: rtc/src/rtc_ice.c, rtc/src/rtc_ice.h

Implemented

Section	Feature	Notes
5.1.1	Host candidate gathering	Interface enumeration via `getifaddrs` (Linux/macOS) / `GetAdaptersAddresses` (Windows)
5.1.1	Server-reflexive (SRFLX) candidate gathering	STUN Binding Request to configured STUN server
5.1.2.1	Candidate priority calculation	`priority = (type_pref << 24) + (local_pref << 8) + (256 - component_id)`
7	Connectivity checks via STUN Binding Requests	With USERNAME, PRIORITY, ICE-CONTROLLING, USE-CANDIDATE
7	STUN Binding Response generation	For incoming connectivity checks
—	ICE credential generation	Random `ufrag` and `pwd`
—	State machine	NEW → GATHERING → CHECKING → CONNECTED / FAILED / CLOSED
—	ICE-controlling role	Always controlling with tie-breaker
—	Remote credential and candidate storage	From SDP parsing and trickled candidates
RFC 8863	Trickle candidate ingestion	`rtc_peer_connection_add_ice_candidate()` parses and stores remote candidates
RFC 8839	`a=ice-options:trickle`	Emitted and parsed

Not Implemented

Section	Feature	Impact
5.1.1.2	Relay (TURN) candidate gathering	TURN client exists but not wired into ICE gather
5.1.3	Full local trickle gathering	Local candidates are still emitted synchronously today
7.3	ICE-controlled role	Always controlling; no role negotiation with peer
7.2.5.3	Triggered checks	Not implemented
8	Regular nomination	Simplified: first successful check wins
8	Aggressive nomination	Not explicitly supported
6	Candidate pair formation / checklist ordering	Simplified: ordered by priority, not full pair algorithm
7.2.5.2.2	Peer reflexive candidate discovery	Not implemented
11	ICE keepalives (STUN Binding Indications)	Not implemented
12	ICE restart after connection	Restart is supported only before connection start
—	Multiple components (RTP + RTCP)	Single component only (RTCP-mux assumed)
—	IPv6 candidate gathering	Not implemented
—	`a=end-of-candidates` signaling	End-of-candidates is signaled via NULL callback, not SDP attribute

Known Limitations

ICE checklist, nomination, role conflict, and consent freshness remain simplified.

RFC 8829 — JSEP

"JavaScript Session Establishment Protocol"

Files: client/src/rtc_peer.c, client/include/rtc/rtc_peer.h

Implemented

Feature	Notes
`createOffer()`	Generates SDP from transceivers + ICE + DTLS state
`createAnswer()`	Generates answer SDP after remote offer is set
`setLocalDescription()`	Applies local SDP, starts ICE gathering
`setRemoteDescription()`	Applies remote SDP, starts connection if both set
`addTrack()`	Creates transceiver with sendrecv direction
`removeTrack()`	Marks sender inactive and updates transceiver direction before connection start
`addIceCandidate()`	Adds trickle ICE candidate
Signaling state machine	stable → have-local-offer → stable (on answer)
Automatic ICE+DTLS+SRTP after both descriptions set	Connection starts on the runtime logical transport

Not Implemented

Feature	Notes
`restartIce()` after connection	Pre-connect restart exists; post-connect ICE restart is not supported
Rollback (set local desc type="rollback")	Not supported
`getTransceivers()` returns copy	Returns direct pointers
Renegotiation (subsequent offer/answer)	Not supported
`RTCRtpTransceiver.stop()`	Not implemented
Unified Plan compliance	Basic transceiver model, not full Unified Plan

RFC 8834 — Media Transport and Use of RTP

"Media Transport and Use of RTP in WebRTC"

Files: client/src/rtc_peer.c, rtc/src/rtc_rtp.c

Implemented

Feature	Notes
RTP/RTCP multiplexing on single port	Via `a=rtcp-mux` and RFC 7983 demux
SRTP mandatory encryption	All RTP/RTCP protected via SRTP/SRTCP
BUNDLE (single transport for all media)	All m= lines share one ICE/DTLS/SRTP transport
Dynamic payload types	Negotiated via SDP

Not Implemented

Feature	Notes
RTP header extensions	`transport-cc` and abs-send-time helpers exist; `mid` and audio-level are not wired
Reduced-size RTCP	Always full-size SR/RR
Bandwidth adaptation based on RTCP	GCC/Transport-CC implemented; encoder auto-wiring remains app-owned
SSRC multiplexing per spec	Per-transport SSRC → stream demux (`rtc_rtp_demux`), shared by peer + SFU

RFC 8839 — ICE SDP Attributes

"Session Description Protocol (SDP) Offer/Answer Procedures for Interactive Connectivity Establishment (ICE)"

Files: rtc/src/rtc_sdp.c

Implemented

Feature	Notes
`a=ice-ufrag:`	Per-media section
`a=ice-pwd:`	Per-media section
`a=candidate:`	Full candidate line formatting with foundation, component, protocol, priority, address, port, type
`a=fingerprint:`	SHA-256 fingerprint
`a=setup:`	DTLS role (actpass for offerer, active for answerer)

Not Implemented

Feature	Notes
`a=ice-options:trickle`	Emitted and parsed
`a=ice-lite`	Not supported
`a=end-of-candidates`	Not signaled
`a=remote-candidates`	Not supported

RFC 8888 — Transport-CC

"RTP Control Protocol (RTCP) Feedback for Congestion Control"

Implemented for the draft-holmer transport-wide CC wire format used by the stack.

Feature	Status
Transport-wide sequence numbers	✅ Implemented
`abs-send-time` RTP header extension helper	✅ Implemented helper
RTCP Transport Feedback packets	✅ Build + parse implemented
Delay-based bandwidth estimation	✅ Simplified GCC trendline estimator
Google Congestion Control (GCC)	✅ Delay + loss controller implemented

Other Relevant Standards (Not Explicitly Referenced)

Standard	Topic	Status	Successor
RFC 8831	WebRTC Data Channels	⚠️ Custom wire protocol over DTLS; not SCTP-based per spec	Current
RFC 4960	SCTP	❌ Not implemented (data channels use custom DTLS framing)	Obsoleted by RFC 9260
RFC 8261	SCTP over DTLS	❌ Not implemented	Current
RFC 8826	WebRTC Security Architecture	⚠️ DTLS+SRTP used; no consent freshness, no SRTP replay protection	Current
RFC 7675	STUN Consent Freshness	❌ Not implemented	Current
RFC 3264	SDP Offer/Answer Model	⚠️ Basic offer/answer works; no renegotiation	Updated by RFC 6157, 8843
RFC 5245	ICE (obsoleted by 8445)	N/A — Uses RFC 8445	Obsoleted by RFC 8445

RFC Migration Plan

Roadmap for updating mrtc code from current (older) RFC implementations to their successors, ordered by priority.

Phase A: Critical Security & Interop (do first)

A1 — SRTP Replay Protection (RFC 3711 §3.3.2)

Why now: Security-critical gap; captured packets can be replayed.
Work: Add 64-packet sliding window replay list in rtc_srtp.c. Check packet index against window before decryption.
Files: rtc/src/rtc_srtp.c, rtc/src/rtc_srtp.h
Effort: Small

A2 — STUN → RFC 8489 Long-Term Auth v2

Why now: RFC 5389 is obsoleted. Modern TURN servers (coturn 4.6+) prefer SHA-256 auth. Current MD5 long-term auth has known weaknesses.
Work:
- Add PASSWORD-ALGORITHMS and PASSWORD-ALGORITHM attribute support in rtc_stun.c
- Implement SHA-256 based HMAC key derivation alongside MD5
- Add USERHASH attribute (optional, privacy feature)
- Support NONCE with STUN-COOKIE prefix for nonce rotation detection
Files: rtc/src/rtc_stun.c, rtc/src/rtc_stun.h
Effort: Medium

A3 — TURN → RFC 8656

Why now: RFC 5766 is obsoleted. Depends on A2 (RFC 8489 auth).
Work:
- Add REQUESTED-ADDRESS-FAMILY attribute for dual-stack (IPv4+IPv6)
- Support ADDITIONAL-ADDRESS-FAMILY for dual allocation
- Wire TURN relay candidates into ICE gathering (rtc_ice.c)
- Add allocation/permission/channel refresh timers
- Fix TURN auth key truncation bug (pass lt_key as uint8_t* with explicit length, not const char*)
Files: rtc/src/rtc_turn.c, rtc/src/rtc_turn.h, rtc/src/rtc_ice.c
Effort: Large

Phase B: RTCP & Feedback (aligns with ROADMAP Phase 2)

B1 — RTCP Feedback: NACK/PLI/FIR (RFC 4585 + RFC 5506)

Why now: Required for video quality and browser interop.
Work:
- Implement Generic NACK (PT=205, FMT=1), PLI (PT=206, FMT=1), FIR (PT=206, FMT=4)
- Add a=rtcp-fb: SDP generation and parsing
- Support reduced-size RTCP (RFC 5506) — send FB without compound SR/RR+SDES
- Add NACK retransmit buffer (512 packets) in sender
- Wire PLI → video_encoder_request_keyframe()
Files: rtc/src/rtc_rtcp.c, rtc/include/rtc/rtc_rtcp.h, client/src/rtc_peer.c, rtc/src/rtc_sdp.c, rtc/include/rtc/rtc_sdp.h
Effort: Large

B2 — RTCP Compound Packets & SDES CNAME (RFC 3550 §6.1 + RFC 7022)

Why now: Spec requires SR/RR+SDES in every compound packet. RFC 7022 updates CNAME generation.
Work:
- Generate compound RTCP (SR/RR + SDES CNAME) instead of bare SR/RR
- Generate CNAME per RFC 7022 (random, persistent per session)
- Implement bandwidth-adaptive RTCP interval (replace fixed 5s timer)
Files: rtc/src/rtc_rtcp.c, client/src/rtc_peer.c
Effort: Medium

B3 — RTP/RTCP Mux Compliance (RFC 5761)

Why now: Already using single port, but should validate per RFC 5761 demux rules.
Work: Verify RTCP PT range demux in rtc_transport.c matches RFC 5761 §4. Already mostly done via RFC 7983.
Files: rtc/src/rtc_transport.c
Effort: Small (audit only)

Phase C: ICE Improvements

C1 — Full Trickle ICE Gathering (RFC 8863)

Why now: All browsers use trickle ICE. Current all-in-SDP approach adds latency.
Work:
- Emit local candidates incrementally as gathering discovers them
- Add a=end-of-candidates signaling
- Handle dynamic checklist updates as remote candidates arrive
Files: rtc/src/rtc_ice.c, client/src/rtc_peer.c, rtc/src/rtc_sdp.c
Effort: Large

C2 — ICE Controlled Role (RFC 8445 §7.3)

Why now: Required for interop when mrtc is the answerer.
Work:
- Add ICE-CONTROLLED attribute support (already defined in rtc_stun.h)
- Implement role determination from SDP offer/answer (offerer=controlling, answerer=controlled)
- Handle role conflict (tie-breaker comparison, 487 error)
- Implement regular nomination (controlled side waits for USE-CANDIDATE)
Files: rtc/src/rtc_ice.c, client/src/rtc_peer.c
Effort: Medium

Phase D: Bandwidth Estimation (aligns with ROADMAP Phase 3)

D1 — RTP Header Extensions (RFC 8834, RFC 5285)

Why now: Required for transport-cc and abs-send-time.
Work:
- Implement one-byte header extension format (RFC 5285)
- Add a=extmap: SDP generation and parsing
- Support abs-send-time, transport-cc seq number, mid, audio-level
Files: rtc/src/rtc_rtp.c, rtc/src/rtc_sdp.c, rtc/include/rtc/rtc_rtp.h
Effort: Medium

D2 — Transport-CC (RFC 8888)

Why now: Needed for delay-based bandwidth estimation.
Work:
- Sender: attach per-packet transport-wide sequence numbers via RTP header extension
- Receiver: track arrival times, build RTCP Transport Feedback packets
- Sender: compare send/recv deltas for delay-based estimation
Files: rtc/src/rtc_rtcp.c, client/src/rtc_peer.c, new rtc/src/rtc_bwe.c
Effort: Large

D3 — GCC (Google Congestion Control)

Why now: Replace AIMD with proper delay-based + loss-based estimator.
Work: Implement GCC algorithm using Transport-CC feedback. Replace rtc_rate_control.c.
Files: new rtc/src/rtc_bwe.c, rtc/src/rtc_rate_control.c
Effort: Large

Phase E: SDP & Protocol Refresh (low urgency)

E1 — SDP → RFC 8866

Why now: RFC 4566 is obsoleted, though changes are editorial.
Work: Audit SDP generation/parsing against RFC 8866 ABNF. Remove any k= line handling. Update doc references.
Files: rtc/src/rtc_sdp.c
Effort: Small

E2 — SRTP Header Extension Encryption (RFC 6904)

Why now: Browsers encrypt mid and audio-level header extensions by default.
Work: After implementing header extensions (D1), add encryption for signaled extensions in SRTP protect/unprotect.
Files: rtc/src/rtc_srtp.c
Effort: Medium

E3 — DTLS 1.3 (RFC 9147)

Why now: DTLS 1.2 remains interoperable; DTLS 1.3 is optional.
Work: Wait for OpenSSL DTLS 1.3 support to stabilize (OpenSSL 3.2+). Then update rtc_dtls.c to negotiate DTLS 1.3 when available, falling back to 1.2.
Files: rtc/src/rtc_dtls.c
Effort: Medium (mostly OpenSSL API changes)

Phase F: Forward Error Correction (aligns with ROADMAP Phase 4)

F1 — UlpFEC (RFC 5109)

Why now: Important for lossy networks. No successor RFC — still current.
Work: XOR-based FEC encode/decode, SDP a=rtpmap:XX ulpfec/90000 negotiation.
Files: new rtc/src/rtc_fec.c, rtc/src/rtc_sdp.c
Effort: Large

Migration Dependency Graph

A1 (SRTP replay)     ─── standalone, do first
A2 (STUN→8489)       ─── standalone
A3 (TURN→8656)       ─── depends on A2
B1 (RTCP feedback)   ─── standalone
B2 (RTCP compound)   ─── standalone
B3 (RTP/RTCP mux)    ─── standalone (audit)
C1 (Trickle ICE)     ─── standalone
C2 (ICE controlled)  ─── standalone
D1 (RTP extensions)  ─── prerequisite for D2, E2
D2 (Transport-CC)    ─── depends on D1
D3 (GCC)             ─── depends on D2
E1 (SDP→8866)        ─── standalone
E2 (SRTP hdr enc)    ─── depends on D1
E3 (DTLS 1.3)        ─── standalone (wait for OpenSSL)
F1 (UlpFEC)          ─── standalone

Effort Summary

Phase	Items	Total Effort	Key Benefit
A	3	Small + Medium + Large	Security, modern TURN interop
B	3	Large + Medium + Small	Video quality, spec compliance
C	2	Large + Medium	Connection speed, browser interop
D	3	Medium + Large + Large	Bandwidth adaptation
E	3	Small + Medium + Medium	Spec freshness, header privacy
F	1	Large	Loss resilience

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RFC_COMPLIANCE.md

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RFC_COMPLIANCE.md

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mrtc — RFC Compliance Report

Summary

RFC 3550 — RTP/RTCP

Implemented

Partially Implemented

Not Implemented

RFC 3711 — SRTP

Implemented

Not Implemented

RFC 7714 — SRTP AES-GCM

Implemented

Not Implemented

RFC 4566 — SDP

Implemented

Not Implemented

Known Limitations

RFC 4585 — RTCP Feedback

RFC 5109 — UlpFEC

RFC 5389 — STUN

Implemented

ICE-Specific Attributes (RFC 8445)

TURN Attributes (RFC 5766)

Not Implemented

Known Bug

RFC 5764 — DTLS-SRTP

Implemented

Not Implemented

RFC 5766 — TURN

Implemented (Client)

Not Implemented

Known Bugs

RFC 6347 — DTLS 1.2

Implemented

Not Implemented

RFC 6455 — WebSocket

Implemented

RFC 7587 — Opus RTP Payload

Implemented

Not Implemented

RFC 7741 — VP8 RTP Payload

Implemented

Not Implemented

RFC 7983 — Multiplexing STUN/DTLS/RTP/TURN

Implemented

RFC 8445 — ICE

Implemented

Not Implemented

Known Limitations

RFC 8829 — JSEP

Implemented

Not Implemented

RFC 8834 — Media Transport and Use of RTP

Implemented

Not Implemented

RFC 8839 — ICE SDP Attributes

Implemented

Not Implemented

RFC 8888 — Transport-CC

Other Relevant Standards (Not Explicitly Referenced)

RFC Migration Plan

Phase A: Critical Security & Interop (do first)

A1 — SRTP Replay Protection (RFC 3711 §3.3.2)

A2 — STUN → RFC 8489 Long-Term Auth v2

A3 — TURN → RFC 8656

Phase B: RTCP & Feedback (aligns with ROADMAP Phase 2)

B1 — RTCP Feedback: NACK/PLI/FIR (RFC 4585 + RFC 5506)

B2 — RTCP Compound Packets & SDES CNAME (RFC 3550 §6.1 + RFC 7022)

B3 — RTP/RTCP Mux Compliance (RFC 5761)

Phase C: ICE Improvements

C1 — Full Trickle ICE Gathering (RFC 8863)

C2 — ICE Controlled Role (RFC 8445 §7.3)

Phase D: Bandwidth Estimation (aligns with ROADMAP Phase 3)

D1 — RTP Header Extensions (RFC 8834, RFC 5285)

D2 — Transport-CC (RFC 8888)