Operator questions
Open Voice Shield FAQ
Open Voice Shield is an additional SIP hop that carries your media. It authorizes the INVITE by source IP, relays and records the RTP through our media server, routes the call on to your own destination, and produces a fraud verdict after hangup. Below: media path, failure behaviour, codecs, latency, storage and data handling.
Call path
The media and signalling path
Does Open Voice Shield sit in the media path?
Yes. Our media server relays the RTP in both directions and captures it; that capture is the recording. There is no mode in which OVS analyzes a call without carrying its media.
Is it a B2BUA, and will it work with mine?
It is a proxy, not a B2BUA. The dialog is authorized, record-routed and relayed onward, and the far end sees your own From and To. To a B2BUA or SBC, OVS is just the next SIP hop, so it works in front of one or behind one.
What happens if Open Voice Shield is unreachable?
If your switch cannot reach the ingress, the INVITE times out and your own failover route takes the call, as it would for any carrier. If the ingress is up but its API is not, the call is answered 503, so your switch fails over immediately instead of waiting.
Which codecs are supported?
Media is passed through, but only G.711 PCMU and PCMA are decoded for transcription. A call in G.722, Opus or a dynamic payload type is relayed and billed normally, and records unsupported_codec instead of a transcript. Offer G.711 on the leg you send us.
Do you support SRTP or encrypted media?
No. The media leg must be plain RTP: encrypted media cannot be captured and decoded for transcription. The onward SIP leg can use TLS, and the portal and API are HTTPS.
How much latency does it add to the call?
One extra SIP hop and one extra media relay: the network time to and from our node, plus the relay itself. No analysis runs during the call, so the fraud pipeline cannot delay live audio.
Which ports do I need to open?
Outbound to the ingress on 5060 UDP or TCP for signalling, and RTP to the advertised media address on the published UDP range. Return media can arrive from any far-end address, so do not restrict the RTP range by source.
The verdict
Timing and enforcement
When is the verdict available?
After the call ends. Decode, transcription and analysis run as an asynchronous job, so the verdict appears on the call detail seconds after hangup. Measured end-to-end timings are on the benchmark page.
Can it block a fraudulent call while it is happening?
No. The analysis exists only once there is a completed conversation to read. The enforcement surface is the next call from that number, which an automation rule can block or divert for a window you choose.
What happens at hangup?
The BYE ends the dialog. Deleting the media session flushes the capture, and the call end is reported with its duration, SIP code and reason. That report starts the decode, transcription and analysis pipeline.
How accurate is it?
We do not publish an accuracy, precision or recall figure: none has been computed against a labelled ground-truth corpus. What we do publish is measured — end-to-end latency, per-stage timing and agreement between analysis tiers on the same calls — on the benchmark page.
Access and data
Authentication, storage and retention
How does it authenticate my switch?
By source IP against a per-tenant CIDR allow-list, checked on the initial INVITE. There is no registration and no digest credential. REGISTER is answered 403: OVS is not a registrar.
Where are recordings stored?
On the operator's own disk, or in an operator-controlled S3 bucket with server-side encryption and public access blocked. Nothing is written to storage the operator does not own.
How long is call data kept?
Recordings, media captures and SIP logs each have their own retention window. Expiry deletes the object; it is not a flag. Call detail records, transcripts, analyses and the audit log have no automatic deletion today.
What data leaves the server?
Speech-to-text runs either on the operator's own server, where the audio never leaves it, or against a cloud API. Fraud analysis sends the transcript and call metadata to a cloud analysis API. Each call's detail page shows whether each stage ran locally or in the cloud.
Can I get the pcap and the SIP ladder for a call?
Yes. Each call keeps its media capture and its traced SIP messages. Both download from the call detail, and both are in the offline analyst bundle with the report, the transcript and the recording.
What happens when my prepaid balance runs out?
The next call is refused at authorization with SIP 402. A call already up is capped by a dialog timeout computed from the remaining balance, so it cannot outrun its funding. During a pilot, balance gates are skipped; security checks still apply.