examples

rvoip examples

Start here. These are runnable, scenario-oriented examples for building with rvoip — organized by what you want to build, not by which API you use. Each is a standalone Cargo project with its own README and (for multi-process demos) a ./run_demo.sh that boots every process and checks the result.

Maturity scope

Examples 01-10 target rvoip-sip, the beta-qualified product — the only workspace product covered by the SIP release gate. Examples 11-14 demonstrate available developer-preview products: the AI harness and vCon path, cross-transport WebRTC-to-SIP escalation, Amazon Connect integration, and hosted Vapi voice agents.

Beta media defaults to PCMU/PCMA; the fully integrated optional G.729A/G.729AB and AMR-NB/AMR-WB paths are developer preview and are not exercised here. Transports are UDP (interop-tested) and TCP/TLS (supported), with SDES-SRTP in the qualified SIP media envelope. Opus/G.722 are developer-preview additions. DTLS-SRTP, ICE, external TURN configuration, and WebRTC are available as developer previews outside the SIP beta gate. The source of truth is crates/sip/rvoip-sip/docs/COMPATIBILITY_MATRIX.md.

Recommended path

1. 01-quickstart-p2p — your first SIP call.
2. 02-softphone-audio — add real PCMU media.
3. Then jump to whatever you're building below.

The examples

#	Example	Scenario	API surface	Run
01	quickstart-p2p	Minimal peer-to-peer call	StreamPeer	./run_demo.sh
02	softphone-audio	Bidirectional PCMU media (verified)	Endpoint + audio	./run_demo.sh
03	register-to-pbx	REGISTER + call via a PBX	Endpoint	cargo run (env-driven)
04	call-control	Hold / resume / DTMF	SessionHandle	./run_demo.sh
05	blind-transfer	3-party REFER transfer	SessionHandle	./run_demo.sh
06	attended-transfer	Consult + REFER w/ Replaces	SessionHandle	./run_demo.sh
07	secure-call-srtp	Mandatory SDES-SRTP	Config SRTP	./run_demo.sh
08	tls-transport	SIP over TLS (sips:)	Config TLS	./run_demo.sh (needs openssl)
09	ivr-server	Reactive inbound server	CallbackPeer	./run_demo.sh
10	call-center-b2bua	B2BUA bridge + routing	UnifiedCoordinator + server::b2bua	./run_demo.sh
11	ai-harness-demo	Fake ASR/TTS/dialog + vCon evidence	rvoip-harness	cargo run
12	customer-escalation-sip-webrtc	Browser WebRTC chat escalates to Alice's SIP phone	rvoip::app gateway API	cargo run -- --auto-proof
13	sip-to-amazon-connect	SIP headers become Amazon Connect attributes with a live audio bridge	rvoip-amazon-connect	cargo run
14	vapi-agent	One server accepts SIP or WebRTC callers and attaches a Vapi voice agent	rvoip::app + rvoip::vapi	cargo run -- --transport sip|webrtc

Conventions

• Self-contained projects. Each example is its own Cargo workspace and uses local rvoip crates from this checkout through path. The paired version tracks the unified workspace train (0.3.8). When copying an example into your own project, drop path and select the published version you intend to use.
• ./run_demo.sh builds release binaries, boots every process with port readiness checks, prints the combined logs, and exits non-zero on failure. Logs land in each example's logs/.
• RUST_LOG controls stack tracing (info, debug).

Looking for the API reference?

These scenario examples are the productized, multi-process front door. For per-API-surface reference examples (one lane each for endpoint, stream_peer, callback_peer, unified, plus protocol regression fixtures and PBX interop), see the in-crate suite: crates/sip/rvoip-sip/examples/. Each example here notes the in-crate example it was built from.