Wholesale SIP Termination Carrier-Grade Global Routes

Scale your International calling instantly. Connect your platform to our secure network for crystal-clear, reliable calls to any number on the planet. We provide the PSTN connectivity you need to grow.

TW-SIP-T Dashboard with Graphs

Network Performance

Real-time analytics

Live
12.8K
Active Calls
190
Countries
99.97%
Success
<45ms
Latency

Regional Performance

North America
4.2K
↑ 12.5%
Europe
3.8K
↑ 8.3%
Asia Pacific
5.1K
↑ 15.7%
Network Load 68%
Quality Score 99.97%
TW-SIP-T What is SIP Section

What Exactly Is SIP Termination?

SIP Termination connects your internet (VoIP) phone system to the traditional telephone network (PSTN). It 'terminates' your digital calls to any landline or mobile, enabling outbound calling.

SIP Termination

Your VoIP System

Your internet-based phone platform initiates an outbound call

SIP Termination

Twiching routes your call through our global network

PSTN Network

Call reaches any landline or mobile number worldwide

TW-SIP-T Comparison Section
Common Confusion

The Common Mix-Up: Termination vs. Trunking

It's easy to get these two confused, but they work as a team. Here's a simple breakdown of what you need to know.

Trunking

The Connection

This is the digital "pipe" or connection from your phone system to the internet. It gives you the capacity to make and receive calls.

  • Establishes the digital pathway

  • Provides call capacity

  • Handles inbound & outbound traffic

Unlimited
Channels
Instant
Setup

Termination

The Delivery

This is the service that takes an outbound call from your trunk and delivers it to its final destination on the global phone network.

  • Routes calls to destination

  • Connects to PSTN network

  • Ensures call completion

190+
Countries
99.9%
Success Rate
TW-SIP-T Call Quality Section

How We Deliver Unmatched Call Quality

Our intelligent global network is built for one thing: reliability. We constantly monitor routes to find the clearest, fastest, and most cost-effective path for every call.

< 50ms
Avg Latency

Intelligent Routing

AI-powered path selection finds the optimal route for every call in real-time

99.9%
Uptime SLA

Geo-Redundant Network

Multiple data centers ensure failover protection and continuous availability

24/7
Support

24/7/365 Monitoring

Real-time network surveillance detects and resolves issues before they impact you

HD
Voice Quality

Proactive Quality Checks

Continuous testing and optimization maintains crystal-clear audio quality

Why Choose Us

Benefits for Your Business, Instantly

Transform your communication infrastructure with enterprise-grade SIP termination that delivers results from day one.

Most Popular

Global A-Z Coverage

Reach 195+ countries with premium direct routes. No middlemen, no quality loss.

190+
Countries
100%
Direct Routes
Best Value

Reduce Call Costs

Cut your wholesale VoIP expenses by up to 40% while maintaining crystal-clear quality.

40%
Cost Savings
30 days
ROI Timeline
Fast Setup

Instant Scalability

Scale from 1 to 10,000+ concurrent calls instantly. Zero infrastructure changes needed.

Unlimited
Max Concurrent
5 min
Setup Time
Secure

Enterprise Security

Bank-grade TLS 1.3 encryption with advanced fraud detection and prevention systems.

TLS 1.3
Encryption
Real-time
Fraud Detection
Lightning Fast

Ultra-Low Latency

Experience sub-50ms average latency worldwide with our optimized global network.

<45ms
Avg Latency
<5ms
Jitter
Always Here

24/7 Expert Support

Get dedicated support from telecom experts who understand your business needs.

<15min
Response Time
24/7/365
Availability
Who We Help - SIP Termination

Who We Help

Carrier-grade voice termination for businesses that demand reliability, scalability, and exceptional quality.

Call Centers & BPOs

Enterprise Contact Solutions

Empower your agents with crystal-clear HD voice quality and zero-latency connections. Scale seamlessly from 10 to 10,000 concurrent calls without compromising quality.

HD Voice Quality
Opus Codec
Zero Latency
<40ms RTT
Concurrent Calls
Unlimited
Global Coverage
190+ Countries
99.99%
Uptime SLA
HD+
Quality Tier
24/7
Support

Live Performance Dashboard

Real-time metrics and analytics

Live
00:00 12:00 24:00
Call Volume
12.8K
+12%
Avg Quality
98.5%
+2.3%
Response Time
42ms
-8%
Cost/Min
$0.004
-15%

SaaS & CPaaS Platforms

Developer-First Integration

Build voice capabilities into your application in minutes. Our RESTful API and comprehensive SDKs make integration seamless with full white-label support.

API Response
<100ms
Integration Time
5 minutes
Daily Requests
10M+
Security
SOC 2 Type II
99.99%
Uptime SLA
Enterprise
Quality Tier
Dedicated
Support

Live Performance Dashboard

Real-time metrics and analytics

Live
00:00 12:00 24:00
API Calls
1.2M
+18%
Avg Latency
38ms
-5%
Success Rate
99.8%
+0.5%
Active Users
45K
+12%

Global Enterprises

Unified Communications

Consolidate your worldwide voice infrastructure under one reliable provider. Enjoy simplified billing, volume discounts, and dedicated enterprise support.

Cost Savings
Up to 40%
Data Centers
50+ Locations
SLA Guarantee
99.95%
Account Manager
Dedicated
99.95%
Uptime SLA
Premium
Quality Tier
Priority
Support

Live Performance Dashboard

Real-time metrics and analytics

Live
00:00 12:00 24:00
Global Offices
85
+5%
Monthly Savings
$42K
+15%
Integration Time
3 days
-40%
Support Response
15min
-25%

Ready for a Clearer, More Reliable Connection?

Stop worrying about call quality and high costs. Test our network with a free trial and see why businesses trust us to power their global conversations.

Frequently Asked Questions

Everything you need to know about Twiching AI

SIP Trunking is the "pipe" that connects your system to the internet. SIP Termination is the service that takes your outbound calls from that pipe and delivers them to the final landline or mobile number.
Our network supports A-Z termination to over 190 countries. You get premium, direct routes for crystal-clear connections to virtually any destination.
Our service is compatible with virtually any SIP-enabled system, including 3CX, Asterisk, FreePBX, and more. You can connect via IP-based authentication or SIP credentials.
uses an intelligent routing engine that scans hundreds of carrier routes in real-time to find the highest-quality, lowest-latency path for your call, avoiding congestion.
Yes. We are fully STIR/SHAKEN compliant, ensuring your calls to the US are properly attested to help improve answer rates and avoid "Spam Likely" flags.
This is a staging enviroment