Quick Start

This guide walks you through your first CallMeter test from signup to results. You will create an account, set up a project, configure a SIP registrar, run a test with real media, and view quality metrics. The entire process takes under five minutes.

All paid plans include a 7-day trial with full access to get started.

Prerequisites

Before you begin, make sure you have the following ready:

• A SIP server to test against. This can be any SIP-capable system: an enterprise PBX, a SIP trunking provider, a hosted UCaaS platform, or even a lab environment. You need the server's domain or IP address and at least two SIP accounts (one to place calls, one to answer them).

• SIP credentials. You need usernames and passwords for at least two SIP accounts on your target server. If you are testing a SIP trunk, your provider supplies these. If you are testing your own PBX, create two test extensions.

• A web browser. CallMeter runs entirely in the browser. Chrome, Firefox, Safari, and Edge are all supported.

Step 1: Create Your Account

1. Navigate to callmeter.io and click Get Started Free.
2. Enter your name, email address, and a strong password.
3. Check your inbox for a verification email and click the confirmation link. If you do not see it within a minute, check your spam folder.
4. After verification, you will be prompted to create your first organization. Enter a name that represents your team or company (for example, "Acme Telecom" or "Network QA Team").
5. Click Create Organization.

You now have an active CallMeter account with the Owner role, which grants full access to all features.

For detailed information about account setup, team invitations, and SSO configuration, see Creating an Account.

Step 2: Create a Project

Projects are the top-level workspace for organizing your tests, registrars, probes, and media files. Most teams create one project per environment (Production, Staging, Lab) or per SIP platform being tested.

1. From the dashboard, click Projects in the left sidebar.
2. Click the New Project button.
3. Enter a project name. For this quickstart, use something like "My First Project" or "SIP Trunk QA".
4. Optionally add a description to help teammates understand what this project tests.
5. Click Create.

Your new project opens automatically and becomes your active workspace.

Step 3: Add a Registrar

A registrar represents the SIP server your test endpoints will register with before placing calls. Think of it as the connection configuration for your target infrastructure.

1. Inside your project, click Registrars in the sidebar.
2. Click New Registrar.
3. Fill in the configuration:

4. Click Save.

Add SIP Accounts

SIP accounts are the credentials that test endpoints use to authenticate with your registrar. You need at least two accounts: one for the caller group and one for the callee group.

1. Open the registrar you just created.
2. Navigate to the SIP Accounts tab.
3. Click Add Accounts.
4. For each account, enter:
   • Username - The SIP authentication username
   • Password - The SIP authentication password
5. Add at least two accounts.
6. Click Save.

Step 4: Create a Test

A test defines the call scenario you want to execute: how many endpoints, how long they call, what media they send, and which registrar they use.

1. Click Tests in the sidebar.
2. Click New Test.
3. Configure the basic settings:

4. Configure the first group (Group A - Callers):

   • Select your registrar
   • Assign one SIP account
   • Under Callee, select "Target Group B" so Group A calls Group B
   • Under Audio, select a codec (PCMA is the safest default)

5. Configure the second group (Group B - Callees):

   • Select the same registrar
   • Assign the other SIP account
   • Under Callee, select "Receive calls" so Group B answers incoming calls

6. Click Create Test.

Step 5: Run the Test

1. On the test detail page, click Run Test.
2. The test enters the Queued status while a worker is allocated.
3. Within seconds, the status changes to Running. You can see a live countdown timer.
4. Endpoints register with your SIP server, establish calls, and begin sending real RTP media.
5. After the configured duration, the test completes and the status changes to Completed.

During execution, you will see a live overview of endpoint statuses:

If all endpoints show Completed, your test ran successfully. If any show Failed, check the error message. Common issues include incorrect SIP credentials, firewall rules blocking SIP or RTP traffic, or the SIP server rejecting registrations. See Common Test Failures for troubleshooting guidance.

Step 6: Analyze Results

Once the test completes, you have access to rich quality metrics for every endpoint in the test.

Test Run Overview

Click on the completed test run to open the results dashboard. The overview shows aggregate statistics across all endpoints:

• Call Success Rate - Percentage of endpoints that completed calls successfully
• Average MOS - Mean Opinion Score across all endpoints (1.0 to 4.5 scale)
• Average Jitter - Mean packet timing variation
• Average Packet Loss - Mean percentage of lost RTP packets
• Average RTT - Mean round-trip time for RTCP packets

Per-Endpoint Metrics

Click on any individual endpoint to drill into its detailed metrics. You will see time-series charts for:

• MOS (Mean Opinion Score) - The overall call quality indicator, calculated per RFC 3550 and the G.107 E-model. A score above 4.0 is excellent, 3.5 to 4.0 is good, below 3.0 indicates significant quality problems.

• Jitter - Variation in packet arrival timing, measured in milliseconds. Low jitter (under 30ms) indicates stable network conditions. High jitter causes audio distortion and video artifacts.

• Packet Loss - The percentage of RTP packets that never arrived at the receiver. Even 1% packet loss can be audible. Above 5% typically causes noticeable degradation.

• Round-Trip Time (RTT) - The time for a packet to travel from sender to receiver and back, measured from RTCP Sender and Receiver Reports. RTT above 150ms can cause conversational overlap.

• Bitrate - The actual media bitrate in kilobits per second, measured separately for send and receive directions.

• NACK Count - Negative acknowledgement requests, indicating the receiver detected packet loss and is requesting retransmission.

• Audio Level - The RTP audio level in dBov, useful for detecting silence, one-way audio, or clipping.

Every metric is recorded separately for the send and receive directions, giving you visibility into both sides of the media path.

What to Do Next

Now that you have run your first test and understand the basics, here are the recommended next steps:

Learn the Platform

• Key Concepts - Understand organizations, projects, registrars, tests, groups, probes, workers, and metrics in depth.
• Supported Codecs - See all supported audio and video codecs with their characteristics and use cases.
• VoIP Terminology - Reference for SIP, RTP, RTCP, and quality metrics terminology.

Run More Advanced Tests

• Creating a Test - Configure multi-group tests with different registrars, codecs, and regions.
• Analyzing Results - Deep dive into time-series analysis, statistical comparisons, and metric correlation.
• Media Files - Upload custom audio and video files for realistic call simulation instead of synthetic tones.

Set Up Monitoring

• Creating a Probe - Configure continuous SIP monitoring with scheduled calls every 5, 15, 30, or 60 minutes.
• Threshold Configuration - Set quality thresholds and get alerted when metrics exceed acceptable ranges.
• Webhooks - Receive real-time notifications when probe health changes.
• Status Pages - Publish public SLA dashboards for customers and stakeholders.

Scale and Automate

• Workers - Deploy on-premise workers for testing behind firewalls or at specific locations.
• API Authentication - Get your API key to automate test execution.
• CI/CD Integration - Gate deployments on VoIP quality thresholds.
• Plans and Pricing - Understand plan limits and upgrade when you need more endpoints or longer test durations.

Common First-Test Issues

If your first test did not go as expected, here are the most common causes:

For more detailed troubleshooting, see:

• Common Test Failures
• SIP Registration Errors
• Poor Quality Metrics