ARCAMUSA VOICE ENGINEERING
Voice engineering guides
Twelve substantive guides covering SIP architecture, Asterisk PJSIP, RTP, response diagnosis, voice security and caller authentication.
Use these guides as evidence-led workflows. Confirm version, topology and peer requirements before changing production systems.
What Is SIP Trunking? Architecture, Call Flow and Deployment
Learn how a SIP trunk connects a PBX or communications platform to another voice network, and why signaling, media, numbering and policy must be designed separately.
Deep guideSIP Trunk vs Hosted PBX: Architecture and Use Cases
Compare who operates the call-control platform, what reaches the customer edge and where security, routing and endpoint responsibilities sit.
Deep guideHow to Configure an Asterisk PJSIP Trunk on an Ubuntu VPS
Build a provider-neutral PJSIP trunk model on Ubuntu using documentation addresses, explicit objects, controlled firewall scope and verifiable CLI checks.
Deep guideAsterisk PJSIP Endpoint, AOR, Auth and Registration Explained
Map each res_pjsip object to its job so inbound identification, outbound authentication and contact location do not become one ambiguous configuration.
Deep guideAsterisk One-Way Audio: NAT, RTP and SDP Troubleshooting
Diagnose one-way audio by following RTP and advertised SDP addresses through NAT instead of changing every PJSIP NAT option at once.
Deep guideSIP 403 Forbidden: How to Find Who Rejected the Call
Locate the hop that generated a 403, correlate the transaction and evaluate identity, authorization and policy evidence without assuming one universal cause.
Deep guideSIP 488 Not Acceptable Here: SDP and Codec Troubleshooting
Read the offer and answer, compare codecs and media security, and find the endpoint that could not accept the proposed session.
Deep guideSIP 503 Service Unavailable: Routing, Capacity and Upstream Diagnosis
Separate local routing, peer reachability, overload and upstream policy when a SIP transaction returns 503 Service Unavailable.
Deep guideAsterisk SIP TLS and SRTP: Secure Signaling and Media Guide
Distinguish TLS signaling from SRTP media and validate certificates, transport selection and encryption negotiation as separate controls.
Deep guideFail2Ban for Asterisk: SIP Brute-Force Protection Without Locking Out Users
Use log-based banning as one security layer while accounting for shared NAT, false positives, observability and safe recovery.
Deep guideHow to Debug SIP Calls with Asterisk, sngrep and tcpdump
Capture only what is needed, correlate a call across Asterisk and the network, protect sensitive headers and turn verbose logging back off.
Deep guideSTIR/SHAKEN Identity Header Explained: PASSporT, Attestation and x5u
Understand the signed PASSporT conveyed by a SIP Identity header, attestation claims, x5u certificates and why a fabricated header proves nothing.
Related engineering resources
Last substantively reviewed: 2026-09-13