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CloudVNO
Programmable Voice

Programmable Voice — SIP Trunking and Call Control

Build voice applications with carrier-grade SIP trunking, call control APIs, STIR/SHAKEN attestation, and global PSTN connectivity.

Key benefits

Ultra-low latency

Sub-200ms call setup times globally with media anchored close to your users across 20+ PoPs worldwide.

STIR/SHAKEN attestation

Full A, B, and C attestation support to ensure your calls reach their destination and display correctly.

Global PSTN connectivity

Reach any phone number in 160+ countries via our direct carrier interconnects and peering agreements.

WebRTC support

Embed browser-based calling directly into your web or mobile app with our WebRTC SDK.

Call recording

Record inbound and outbound calls with dual-channel audio, automatic transcription, and secure storage.

Programmable IVR

Build sophisticated call trees, menus, and routing logic using CloudVNO Markup Language or REST webhooks.

How it works

1
Provision

Get a phone number

Provision a DID, toll-free, or SIP trunk endpoint via API or dashboard. Active within seconds in 160+ countries.

2
Call

Make or receive calls via REST

POST to /v1/calls to initiate outbound calls, or configure a webhook URL to handle inbound calls as they arrive.

3
Control

Handle with webhooks

CloudVNO calls your webhook for each call event — answer, DTMF, recording complete — letting you drive call logic in real time.

Code example

from cloudvno import Client

client = Client("YOUR_API_KEY")

# Initiate an outbound call
call = client.calls.create(
    from_="+14155551234",   # Your CloudVNO number
    to="+12125559876",
    url="https://yourapp.com/webhooks/voice",  # TwiML or CloudVNO Markup
)

print(f"Call SID: {call.sid}")
print(f"Status: {call.status}")  # initiated → ringing → in-progress → completed

# Retrieve call details
call = client.calls.get(call.sid)
print(f"Duration: {call.duration}s")
print(f"Direction: {call.direction}")

Features

Inbound & outbound PSTN calls
Call recording (dual-channel)
Programmable IVR
SIP trunking
STIR/SHAKEN A/B/C attestation
Voicemail detection & transcription
Call forwarding & transfer
Conference calling (up to 250 participants)
WebRTC browser & mobile SDK
Real-time transcription via webhooks
Call analytics & CDR export
Queue management & hold music
Outbound (US)
$0.013
per minute
Inbound (US)
$0.0085
per minute
SIP trunking
$0.008
per minute

Pay as you go. No minimums. No contracts. Volume discounts available.

Get started in 5 minutes

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