Compliance recording isn’t “turn recording on.” It’s being able to answer, months later, questions like: was this specific call recorded? why not? where is it now? can you produce it, and prove it wasn’t altered?
RustPBX’s recording pipeline is built to make those answers routine. Here’s the operational playbook.
Interconnecting with carriers is where good PBX software meets bad SIP. The other end has opinions: it rewrites numbers, rejects your codec order, sends private IPs in SDP, and insists on headers you’ve never heard of. A session border controller exists to absorb that.
When a call misbehaves, the CDR tells you what happened and the media counters tell you how it sounded. Neither tells you exactly what was on the wire. For that you need packet-level signaling — captured continuously, indexed by call, queryable weeks later, and cheap enough to leave on permanently.
That’s SipFlow.
“Outbound calls have no audio, inbound is fine.” If you’ve run SIP for a week, you’ve met this sentence. It’s almost never the codec — it’s address advertisement: the PBX told the far end an IP it can’t reach.
RustPBX separates the three addresses that SIP/WebRTC care about, and lets you override each per deployment, per network profile, or per trunk.
“Record all calls” is how recording requirements start. Then legal arrives: some calls must not be recorded. Finance wants the billing segments separately. The auditor wants retention measured in years, storage in object buckets, and downloads that don’t melt the PBX. Here’s how RustPBX’s recording pipeline grows with you.
Browsers won’t touch your WebRTC endpoints without TLS. Carriers increasingly demand SIP over TLS. And certificates that expire on a Saturday are how outages are born. RustPBX bakes the whole story in.
A PBX with a public IP gets scanned within minutes of going live. SipVicious and its cousins register-probe every open 5060 on the internet, hunting for default passwords they can turn into toll-fraud. Here’s the hardening checklist for the first 48 hours — all of it built in, none of it requiring external tooling.
Wholesale voice is a business run on spreadsheets and margins, so the platform running it should speak spreadsheet: exact route matches, predictable rounding, and a way to answer “why did this call route the way it did” without guesswork. The latest RustPBX wholesale addon update focuses on exactly that.
Debugging SIP calls usually means running tcpdump, downloading PCAP files, and opening Wireshark. It works, but it’s slow, requires server access, and doesn’t scale.
SipFlow is RustPBX’s built-in signaling capture system. It records every SIP message and RTP packet, stores them efficiently, and presents a visual timeline in the web console. Need a recording? Generate a WAV from the captured RTP data with one click. Need to debug a call from last week? Query it by call ID.
And when a single node isn’t enough, SipFlow supports cluster deployment with consistent hashing.
Ride-hailing apps, delivery platforms, and marketplaces all share a common requirement: connect two parties on a phone call without revealing their real numbers. This is the classic privacy number (AX/NXA) use case.
RustPBX’s SBC addon includes a JSON-RPC router that makes this straightforward. Every incoming call can query your backend API, get the real destination, rewrite the numbers, and connect the call – all in real time.
Your team can call each other internally. Now you need to connect to the PSTN through a carrier, let customers call in from web browsers, and manage it all without spinning up a separate SBC appliance.
RustPBX’s SBC addon handles this in a single runtime – no Janus, no FreeSWITCH, no extra media servers.
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