HD voice, by default
Codecs transcoded at the edge when two ends disagree: Opus and G.722 for HD voice, G.711 µ/A-law, G.729, iLBC and GSM for everything else, and T.38 for fax.
- Opus
- G.722
- G.711
- G.729
- iLBC
- GSM
- T.38
Private SIP trunking
Carrier-grade, encrypted SIP trunking for any PBX. Open an account with just an email and a password, pay in Bitcoin, Monero or USDT, and route your first call within minutes.
Works with
Pay with
01 · No-KYC onboarding
That is the entire form.
02 · How a PrivSIP call travels
Asterisk · FreeSWITCH · 3CX · FreePBX · Teams DR
Point any standards-compliant PBX or softswitch at PrivSIP. Register with digest credentials and, if you want, lock registration to an IP allow-list per trunk.
SIP over TLS 1.3 · SRTP (SDES / DTLS)
Signalling and audio are encrypted from your PBX to our edge. Plain UDP exists for legacy hardware, but every new trunk starts on TLS.
Anycast SBCs · 22 PoPs · topology hiding
Session border controllers in 22 cities strip internal headers, absorb SIP floods and fail over in under two seconds. Your trunk registers to the nearest healthy PoP.
Tier-1 only · live ASR / PDD scoring
Least-cost routing that refuses grey routes. Each upstream carrier is scored live on answer-seizure ratio and post-dial delay, and weak paths are drained automatically.
190+ countries · STIR/SHAKEN on US calls
Calls reach the public network with your caller ID intact, whether it's a PrivSIP number or one you hold elsewhere. Routes stay clean because traffic is scored, not because your options are limited.
03 · SIP trunk features
Every PrivSIP plan ships with the same network, codecs and fraud protection. Plans only change how many channels and minutes you get.
Compare plansCodecs transcoded at the edge when two ends disagree: Opus and G.722 for HD voice, G.711 µ/A-law, G.729, iLBC and GSM for everything else, and T.38 for fax.
If your PBX stops answering, calls re-route to a second SIP URI, another trunk or a mobile number. Registration moves to a healthy PoP before the caller notices.
Calls to 190+ destinations, plus every type of DID number, local and toll-free included, in 190+ countries. Pick a DID in the dashboard and it is live in minutes, with no paperwork.
Per-trunk daily spend caps, destination allow-lists, CPS ceilings and IRSF pattern detection. A compromised PBX costs you nothing beyond your cap.
Call records live for 30 days, or 7 if you prefer, and are then deleted rather than archived. Pull them first over the REST API or real-time webhooks.
Median first response on PrivSIP live chat, around the clock. The person who answers can read a SIP trace, and they will not ask for your name.
one-way audio on outbound, inbound fine
Your SDP is advertising 10.0.0.12 — set external_media_address in pjsip.conf and reload.
04 · Connect any PBX
PrivSIP is a standard SIP trunk over TLS. No agent to install, no vendor lock-in, and codecs are transcoded at the edge when two ends disagree.
Setup guide · Open-source PBX
05 · Global SIP network
The PrivSIP core runs in Frankfurt, Amsterdam and Ashburn, with edge SBCs in nineteen more cities. Nothing in the signalling path runs on shared cloud tenancy or on equipment we don't operate.
Nearest PoP is estimated from your browser's time zone. Nothing is sent to us.
06 · Crypto-only billing
More than 250 coins at checkout. We receive a transaction, not a cardholder name, billing address or bank reference, and with no chargebacks there is no fraud premium in your rate.
07 · Plans
Every PrivSIP plan runs on the same encrypted network, with unlimited inbound and a generous bundle of outbound minutes. Tell us how many people make calls and we'll point you at the right one.
The whole office on one trunk.
$120/ month
08 · Data we keep
A provider can only hand over what it has. PrivSIP keeps the minimum a SIP trunk needs to bill and to fight fraud, deletes it on a fixed schedule, and never collects the rest.
09 · Live support
No ticket queue and no scripted first line. PrivSIP live chat goes straight to the people who run the network, and you don't need an account to ask a question.
Today · anonymous visitor
Trunk registered but inbound calls get a 404.
03:14Checking now. Your number routes to your trunk, but the PBX only answers for “s”. Match the inbound context to the number, or add a catch-all.
03:14That was it. Calls are landing, thanks!
03:16Anytime. One more tip: put Opus or G.722 first in your codec list for HD audio. G.711 and G.729 stay as fallbacks.
03:16Popular questions
The six things people ask most often. The help center has 55 detailed answers on billing, numbers, setup and more.
Can't find it? An engineer usually answers in under a minute.
Yes. A PrivSIP account needs an email address and a password, nothing else. We never ask for a name, ID scan, selfie or proof of address to run a SIP trunk. The same goes for phone numbers: any local requirements are handled by PrivSIP, never passed on to you.
Call detail records (time, from, to, duration and cost) for 30 days, or 7 if you choose, because billing and fraud protection depend on them; SMS delivery records follow the same window, without the message text. SIP traces and registration IPs are kept for 72 hours for troubleshooting, and live-chat transcripts for 90 days. PrivSIP never records call audio. When a window ends, the data is deleted rather than archived.
Pick by the number of people making calls at the same time. Solo (5 channels) suits one to five people, Team (12 channels) small offices, Business (30 channels) growing companies, Contact Center (75 channels) support and sales floors, and Enterprise (180 channels, burst to 360) platforms and large organisations. Every plan runs on the same network with the same encryption and codecs.
PrivSIP's crypto checkout supports more than 250 coins and tokens, including Bitcoin, Monero, Ethereum, USDT and USDC on several networks, Litecoin, Solana and Tron. Prices are set in US dollars and the rate is locked for 20 minutes while you pay.
Anything that speaks standard SIP (RFC 3261): Asterisk, FreePBX, FreeSWITCH, FusionPBX, 3CX, Yeastar, Grandstream UCM, VitalPBX, Kamailio and Microsoft Teams Direct Routing through a certified SBC, plus ordinary softphones and desk phones. The PrivSIP dashboard has ready-to-paste config for the most common ones.
PrivSIP stands for freedom of communication. We do not ask for a use-case statement and do not police content: business lines, contact centers, automated and outbound campaigns, reselling under your own brand and privacy-sensitive work are all welcome. The acceptable use policy restricts only attacks on the PrivSIP network and activity meant to harm the carriers and operators that use it.