rvoip-sip 0.3.8

# rvoip-sip Beta Topology Profiles

Date: 2026-07-25

This document defines which deployment shapes beta is allowed to claim and
which shapes remain post-beta or advanced tuning work.

Current reference: [Beta Release Candidate Report](BETA_RELEASE_REPORT.md),
run `20260724T231400Z`, generated from clean tested commit `8d44fb35`.

## Beta-Supported Profiles

| Profile | Status | Required validation |
|---------|--------|---------------------|
| Local loopback app | Supported | In-process examples and integration tests. |
| Basic SIP client | Supported | `Endpoint` and `StreamPeer` outbound call, registration, DTMF, hold/resume. |
| Basic SIP server | Supported | `CallbackPeer` inbound call, reject/accept, DTMF, BYE cleanup. |
| Asterisk PBX | Interop tested | UDP/TLS registration and calls, digest auth, SDES-SRTP where claimed. |
| FreeSWITCH PBX | Interop tested | Mirrors the Asterisk matrix where feasible. |
| Kamailio transaction-stateful proxy | `0.3.8` release gate | Real-process UDP/TCP/TLS, routing, CANCEL, forking, ACK, response, and cleanup matrix. |
| OpenSIPS transaction-stateful proxy | `0.3.8` release gate | Independent real-process execution of the same proxy matrix. |
| SIPp UAC/UAS | Release gate | Standalone load matrix at 30, 100, 300, 1,000, and 2,000 CPS. |
| baresip strict-UA | Interop tested | Strict-UA INVITE, 200 OK, ACK, established call, BYE, and rvoip accept checks. |
| Signaling-only B2BUA/gateway | Supported with limits | Multi-leg signaling tests and clear media relay caveats. |
| Full-media beta perf | Beta target | Media enabled, PCMU/PCMA/DTMF, up to 2,000 CPS in the final clean report. |

## Advanced or Post-Beta Profiles

| Profile | Status | Reason |
|---------|--------|--------|
| Tuned high-CPS above 2,000 CPS | Advanced | Requires explicit tuning, hardware notes, and topology caveats. |
| RTPengine media relay | Investigation | Media-relay integration is separate from the `0.3.8` signaling-proxy conformance claim. |
| Carrier SBC certification | Post-beta | Requires carrier-specific certification and security audit. |
| Browser/WebRTC edge | Post-beta | DTLS-SRTP, ICE, TURN, and browser interop are outside beta. |
| ICE/TURN NAT traversal | Post-beta | Current STUN support is limited address discovery, not ICE. |
| Recording/announcement/IVR media server | Post-beta unless completed | Media-core feature plan still lists gaps. |

## General Full-Media Beta Profile

The default beta performance claim is:

- Media mode: `MediaMode::Enabled`
- Codecs: PCMU (`0`), PCMA (`8`), telephone-event (`101`)
- Optional: comfort noise (`13`) only with `comfort_noise_enabled=true`
- Security: plaintext RTP or tested SDES-SRTP profile
- Target: stepped SIPp/media runs at 30, 100, 300, 1,000, and 2,000 CPS
- Success: the workload-specific ASR threshold, no stuck sessions, RSS slope
  within 15 MB/hour where gated, full application audio-frame delivery, and
  published p50/p95/p99 setup latency
- Soak: the current policy requires the recorded one-hour monolithic and split
  full-media configurations; this is not a 24-hour claim

Results above 2,000 CPS must be labeled as tuned or experimental unless they
use the same general profile and pass the same evidence bar.