SIP termination provider for international voice traffic

Wholesale SIP termination for international voice traffic

CloudyVoice is a SIP termination provider delivering prepaid A–Z international SIP termination across 259 destinations and 19,172 prefix routes, priced at prefix level and published before you register.

What SIP termination is

How a call actually gets from your platform to the called party

SIP termination is the service that takes an outbound call from your platform and completes it to the destination network — it is the "last mile" of a call, not the origination or the phone system itself. A single call moves through a fixed sequence:

Your PBX, dialer or softswitch SIP connection CloudyVoice routing Destination carrier / network PSTN or mobile recipient

Your platform sends a SIP INVITE for the dialled number over your registered or IP-authenticated connection. CloudyVoice looks up the prefix, selects a route to the destination carrier, and hands the call off; the destination network then completes it to the landline or mobile handset. Anyone originating outbound calls in volume needs this layer — call centres and BPOs, PBX and softswitch operators, ITSPs and resellers buying capacity to sell onward, and businesses with international calling traffic that has outgrown a consumer VoIP plan.

A–Z coverage, priced honestly

What "A–Z international termination" means here

A–Z coverage

Fixed and mobile termination across 259 destinations, priced at prefix level rather than a single country average — mobile ranges and premium prefixes are visible before you send traffic.

Prefix-level pricing

Every route is priced by its own prefix, not blended into a country-wide rate. The rate directory publishes these openly, with lowest fixed and mobile figures per destination.

CLI and non-CLI

CLI routes pass calling line identification where the destination network supports it; non-CLI routes do not and are priced accordingly. Both are labelled honestly rather than sold interchangeably.

Billing increments

Every route publishes its increment — 1/1, 60/1 or 60/60 — because on short calls the increment changes effective cost more than the headline rate does.

Authentication

SIP registration (username and password) or static IP authentication, whichever fits your platform.

E.164 dialing

Numbers are dialled in E.164 format — country code plus national number, no local trunk prefix, spaces or brackets.

Before you send volume

Rate decks, testing and traffic review

Current rates for every destination are published on the rate directory; a rate deck matched to your specific destination list and volume is prepared on request. Before committing volume, fund a small prepaid balance and test across several networks in each destination — including busy, invalid and deliberately unanswered numbers — checking two-way audio, caller ID behaviour, post-dial delay and billing accuracy. The route-quality guide sets out the full method, including how to detect false answer supervision. Outbound routes are enabled after traffic review confirms your destinations, expected volume and lawful use case — this exists because non-compliant traffic gets prefixes blocked at carrier level for every customer sharing them, not as paperwork for its own sake.

Buyer reference

Technical specifications

Authentication
SIP registration (username/password) or static IP authentication
Dialing format
E.164 — country code plus national number, no trunk prefix
Billing increments
Published per route (1/1, 60/1 or 60/60) — see each destination’s rate page
Balance model
Prepaid USD wallet; balance does not expire while the account is active
Payment model
Cryptocurrency prepaid — BTC, ETH, USDT (ERC-20 and TRC-20); no monthly commitment
Rate effective dates
Published on every destination page
Route testing
Recommended before volume commitment — see the route-quality guide
Supported platforms
Any SIP-capable softphone, IP-PBX, gateway, dialer or softswitch — including Zoiper, MicroSIP, Linphone, Asterisk, FreeSWITCH, ViciDial, VOS3000 and 3CX
Codecs
Confirmed during onboarding
SIP transport
Confirmed during onboarding
DTMF
Confirmed during onboarding
CDR availability
Full CDR access on a dedicated VOS/softswitch deployment; hosted-account CDR access is confirmed during onboarding
Rate deck format
Confirmed during onboarding
Channels
Confirmed during onboarding
CPS
Confirmed during onboarding
Failover
Available on wholesale and VOS routes; confirmed per destination during onboarding
Related services

Connecting to and buying more of this network

SIP trunking

The connectivity itself — registering or IP-authenticating your PBX, dialer or softswitch to this network.

Wholesale VoIP

The same A–Z network sold at volume, with committed capacity and rate decks matched to your destination mix.

Call center VoIP

SIP termination sized specifically for dialer and call-centre traffic patterns.

Questions

Frequently asked questions

What is SIP termination?

SIP termination is the service that completes an outbound call from your platform to the destination network — the PSTN or a mobile operator. It is distinct from SIP trunking (the connectivity between your platform and the provider) and from the phone system itself.

Who needs SIP termination?

Anyone originating outbound calls in volume: call centres and BPOs, businesses with international calling needs, PBX and softswitch operators, and ITSPs or resellers buying capacity to sell onward.

What is the difference between CLI and non-CLI routes?

A CLI route passes calling line identification so the called party sees a number, where the destination network supports and permits it. A non-CLI route does not present a number and is usually cheaper, but is unsuitable where the recipient needs to identify or call back the caller.

How are channels and CPS handled?

Concurrent channels and calls per second are agreed per account against your expected peak traffic — confirmed during onboarding once you share your expected volume and traffic shape.

Which platforms can connect?

Any SIP-capable softphone, IP-PBX, gateway, dialer or softswitch — including Zoiper, MicroSIP, Linphone, Asterisk, FreeSWITCH, ViciDial, VOS3000 and 3CX. Registration and static IP authentication are both supported.

Is there a minimum commitment?

No. Accounts are prepaid with a $29 minimum top-up and no monthly subscription.

Tell us what you need to carry

Send your destinations, expected daily minutes, traffic type and platform. We will come back with a route recommendation and a test plan.