CalloraVoipSdk 4.8.0

[BechsteinDigital]

CalloraVoipSdk 4.8.0

A stack-wide hardening release, plus mutual TLS per SIP line and a public PCM transcoding surface.
4.8.0 closes a long series of protocol-layer review findings across every wire boundary — DTLS-SRTP,
STUN, TURN, RTP/RTCP, SDP, SIP and the WebRTC media plane — and adds two additive public capabilities.

The security work is defensive throughout: fail-closed decoding, constant-time comparisons, anti-amplification
and anti-slowloris deadlines, source binding on plaintext legs, and bounded wire-DoS caps at every parser and
listener. The public API only grew — PublicApi.approved.txt gained entries and lost none — so an existing
consumer that does not opt into the new configuration surfaces stays behaviour-identical. Every cap sits far
above any real signalling or media, so legitimate traffic is unaffected.

Headline features

• Mutual TLS per SIP line, with a certificate from memory ([P2][SIP/TLS] Mutual TLS pro Line statt client-weit — und Zertifikat aus dem Speicher statt Dateipfad #183). Outbound SIP over TLS/WSS now presents a
  client certificate when one is configured — sent only if the registrar asks for it (RFC 8446 §4.4.2), so
  behaviour against registrars that do not request one is byte-identical. Two lines to the same registrar can
  present different identities: the identity is bound to the line, keyed into the connection pool
  ((transport, addr:port, identity)), and stamped on every request the line originates (REGISTER, INVITE and
  in-dialog requests, MESSAGE, PUBLISH). Configure it in memory via TlsConfiguration.ClientCertificate (an
  X509Certificate2 you own and dispose — the SDK never disposes it), or override per connect boundary with
  ConnectOptions.LineTls. Per-line domain identity and trust stay fail-closed (RFC 5922). See ADR-064.
• Public PCM transcoding surface ([Audio] Öffentliche PCM-Transcoding-Fläche: Opus/G.711/G.722 ↔ PCM16 für serverseitige Konsumenten #205). IAudioPayloadCodec + AudioPayloadCodecFactory (in
  CalloraVoipSdk.Audio.Abstractions) let a server-side consumer transcode Opus / G.711 (A-law/µ-law) / G.722
  ↔ PCM16 without taking a direct dependency on Concentus or NAudio — neither appears in any public signature.
  The prototypical use case is an SFU that decodes N−1 legs to mix a phone participant into a WebRTC conference.
  I/O is PCM16 little-endian byte[]; one instance per stream direction. Shipped transitively via the
  CalloraVoipSdk meta-package — no new PackageReference. See ADR-065.

Security hardening

• DTLS-SRTP: constant-time fingerprint comparison (RFC 8122), zeroed private-identity staging bytes, a
  stateless HelloVerifyRequest cookie before the amplified certificate flight (RFC 6347 §4.2.1), a handshake
  deadline (DtlsHandshakeOptions.HandshakeTimeout, default 20 s), an ordered single-writer egress with real
  error propagation, and — new in this release — post-handshake association servicing: a control-receive
  loop that notices a peer close_notify/alert, ends the association deterministically, and stops media from
  flowing under a keying channel the peer considers closed (RFC 8827 §6.5). See ADR-066.
• STUN/TURN: auth-response integrity binding, a stateless HMAC nonce (amplification cap), TCP/TLS slowloris
  deadlines on both servers, fail-closed ALTERNATE-SERVER/decode handling, and immediate teardown of a surplus
  TURN relay allocation (RFC 8656 §7/§3.9).
• RTP/RTCP wire-DoS caps: RTCP compound budgets, a transport-cc feedback expansion cap, depacketiser frame
  caps (with discard telemetry), and a BUNDLE reception-state cap — an authenticated peer can no longer exhaust
  process memory through compound floods, header-field expansion, unterminated frames or SSRC spray.
• SIP / app-domain: fail-closed ICE (no session without a validated candidate pair), an atomic per-line
  concurrent-call cap, bounded inbound connection/session/transaction admission with slowloris deadlines, and a
  redacted SipCredentials.ToString() (the cleartext password no longer reaches logs).
• STUN wire fail-closed: StunMessageCodec.Decode now rejects a whole malformed message rather than
  accepting a partial one, closing an auth-bypass primitive.

Correctness fixes

• Inbound audio carries its real RTP timestamp (RFC 3550 §5.1) — an SFU no longer stamps forwarded audio at
  0.
• SDP BUNDLE is grouped semantically (RFC 5888/8843/9143), a remote answer is validated against the local
  offer (RFC 3264 §6 / RFC 8829), and the offerer sends the codec the answer accepted (RFC 3264 §6.1).
• DTLS handshake timeouts are classified by wall-clock, so a stalled handshake reports a timeout rather than
  a generic protocol error.

Upgrading

• No breaking change. All new APIs are additive; a fixed configuration that does not use them is unaffected.
• Deprecation: TlsConfiguration.AcceptUntrustedCertificates is now an [Obsolete] alias for
  TrustMode = SipTlsTrustMode.DangerousAcceptAnyChain. It still compiles and behaves as before; migrate to
  TrustMode at your convenience.
• Notable behaviour changes (correct, but observable): a non-conforming STUN server that omits
  MESSAGE-INTEGRITY on a credentialed Binding success now triggers a safe host-only fallback; plaintext legs
  discard foreign RTP/RTCP; a 416 no longer downgrades sips: to sip:; a structurally mismatched SDP answer
  is rejected; and SipCredentials.ToString() prints *** for the password. See CHANGELOG.md for the full
  list.

Status notes

• Full ICE and the browser-facing WebRTC peer remain opt-in and not yet browser-interop-certified — this
  release hardens them but does not change that posture. Validate for your trunk/peer before enabling.
• The DTLS close-block work ([DTLS] Serve the association after key export — peer close_notify, alerts, no_renegotiation #190/[DTLS] Order the egress, propagate send failures, drain close_notify before cancelling #191) is wired for the WebRTC/bundle media plane; the SIP media-session owner
  reaction to a peer close is a documented follow-up (a SIP leg ends via BYE, not a mid-call DTLS close_notify).

See CHANGELOG.md for the concise per-change entry, and
docs/adr/ADR-064-per-line-sip-mutual-tls.md,
docs/adr/ADR-065-public-pcm-transcoding-surface.md and
docs/adr/ADR-066-dtls-post-handshake-association-servicing.md
for the architecture decisions.

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This discussion was created from the release CalloraVoipSdk 4.8.0.