What Is VoIP? The Complete Guide to Voice over IP

If you've made a call on WhatsApp, joined a Zoom meeting or called a business with a modern phone system, you've already used VoIP. It's now the standard way businesses make and receive calls, and the technology behind most of the world's voice traffic.

But what exactly is VoIP, how does it work, and what do you need to know before using it for your business?

This guide covers everything in plain English: how a VoIP call travels, the technology and terms behind it, what affects call quality, what it costs, how to keep it secure, and how to choose the right provider. By the end, you won't need to look anywhere else.

Contents

1. What is VoIP?

2. A short history of VoIP

3. How does VoIP work? (step by step)

4. The technology behind VoIP: SIP, RTP and codecs

5. VoIP vs. traditional landlines

6. Types of VoIP services

7. What do you need to use VoIP?

8. How much internet speed does VoIP need?

9. What affects VoIP call quality?

10. Phone numbers on VoIP: DIDs, toll-free and porting

11. VoIP features

12. Advantages of VoIP

13. Disadvantages of VoIP (and how to solve them)

14. VoIP security and fraud

15. STIR/SHAKEN, robocalls and caller ID

16. Emergency calls (911) on VoIP

17. How VoIP pricing works

18. Wholesale VoIP explained

19. How to choose a VoIP provider

20. Frequently asked questions

1. What is VoIP?

VoIP stands for Voice over Internet Protocol. It's a technology that lets you make and receive phone calls over an internet connection instead of a traditional copper phone line.

In simple terms, VoIP turns your voice into digital data, sends that data over the internet (or a private IP network) in small pieces, and turns it back into sound at the other end.

VoIP is pronounced "voyp" (rhymes with "boy-p") or spelled out as "V-O-I-P." It's also called:

• IP telephony

• Internet calling

• Broadband phone

• Cloud phone or cloud calling

VoIP calls can reach any phone. You can call another VoIP user, a mobile phone or a traditional landline. When a VoIP call needs to reach a regular phone number, the VoIP provider hands it over to the traditional phone network (called the PSTN, or Public Switched Telephone Network) at the right point.

2. A short history of VoIP

• 1970s: Researchers on ARPANET, the early version of the internet, experimented with sending voice over data networks.

• 1995: VocalTec released one of the first consumer internet phone programs, letting two computer users talk over the internet.

• Late 1990s–early 2000s: Standards like H.323 and SIP (Session Initiation Protocol) were created, making VoIP systems from different companies able to work together.

• 2000s: Broadband internet spread, and services like Skype and the first business VoIP phone systems became popular.

• 2010s: Cloud phone systems, mobile apps, and SIP trunking replaced many traditional business phone lines. Carriers moved more and more of their core networks to IP.

• Today: Phone companies around the world are retiring old copper networks. VoIP is no longer the "alternative." It's how modern voice works.

3. How does VoIP work? (step by step)

A traditional phone call opens a dedicated electrical circuit between two phones for the whole call. This is called circuit switching.

VoIP works differently. It uses packet switching, the same method used to send emails and load websites. Your voice is broken into tiny packets of data that travel across the network independently and are put back together at the other end.

Here's what happens during a VoIP call:

Step 1: You dial. You dial a number on an IP phone, a softphone app on your computer or mobile, or a regular phone plugged into an adapter.

Step 2: The call is set up (signaling). Your phone sends a message to your VoIP provider asking to start a call. This is done using a signaling protocol, usually SIP. The provider works out where the number lives and how to reach it, and the destination phone starts ringing.

Step 3: Your voice is converted to digital data. When you speak, the microphone captures your voice as an analog sound wave. A codec (short for coder-decoder) samples that sound thousands of times per second, converts it into digital data and usually compresses it.

Step 4: The data is split into packets. The digital audio is cut into small packets, typically one every 20 milliseconds, which is 50 packets per second in each direction. Each packet gets a "label" with its destination address, its order in the sequence and a timestamp.

Step 5: The packets travel across the network. The packets travel over your internet connection, through your provider's network and, if needed, through other carriers' networks. This is done using a protocol called RTP (Real-time Transport Protocol). Packets can take slightly different paths and arrive at slightly different times.

Step 6: The packets are reassembled. At the other end, a jitter buffer collects the packets, puts them back in the correct order and smooths out small timing differences.

Step 7: The data is turned back into sound. The codec decodes the data back into an audio signal, and the person on the other end hears your voice. All of this happens in a fraction of a second, in both directions at once.

Step 8: The call ends. When either person hangs up, a SIP message ends the session, and the call details (time, duration, destination) are recorded for billing. This record is called a CDR (Call Detail Record).

What if you're calling a regular phone?

If the person you're calling uses a landline or mobile network, your VoIP provider passes the call to a carrier that connects to that network. A device called a media gateway converts the call from IP to the format that network uses. This hand-off is called termination: the call "terminates" on the destination network.

The reverse, receiving calls from the regular phone network onto VoIP, is called origination.

4. The technology behind VoIP: SIP, RTP and codecs

You don't need to be an engineer to use VoIP, but knowing these terms will help you understand what providers are offering.

SIP (Session Initiation Protocol)

SIP is the most widely used signaling protocol in VoIP. It's the "call control" language that phones and servers use to:

• Start a call (an INVITE message)

• Ring the other phone

• Answer, hold, transfer, or end a call

• Register a phone with the provider so it can receive calls

SIP handles the setup and control of the call. It doesn't carry the voice itself.

RTP (Real-time Transport Protocol)

RTP carries the actual audio (and video) once the call is connected. It adds sequence numbers and timestamps so the receiving side can play the audio back in the right order and at the right speed. SRTP (Secure RTP) is the encrypted version.

Codecs

A codec compresses and decompresses audio. Different codecs balance sound quality against how much bandwidth they use:

Both sides of a call have to agree on a codec. If they don't support the same one, the call is transcoded (converted from one codec to another), which can slightly reduce quality and add delay.

Other terms you'll hear

• PBX (Private Branch Exchange): the phone system inside a business that connects internal extensions and routes calls. An IP PBX is a VoIP version. A hosted PBX or cloud PBX is run by the provider in the cloud.

• SBC (Session Border Controller): a security and traffic-management device that sits at the edge of a VoIP network, like a firewall built specifically for voice.

• WebRTC: technology that lets you make voice and video calls directly in a web browser, with no app to install.

5. VoIP vs. traditional landlines

6. Types of VoIP services

"VoIP" covers several different kinds of services. Knowing which one you need is the first step to choosing the right provider.

Consumer apps

WhatsApp, FaceTime Audio, Skype and similar apps let people call each other free over the internet. They're great for personal use but aren't built for business needs like call routing, call recording or high call volumes.

Hosted PBX / cloud phone systems

The provider runs the whole phone system in the cloud. Your team gets phone numbers, extensions, voicemail, call routing and other features through IP phones or apps. This is the most common choice for small and mid-sized businesses. When it's bundled with video, chat and messaging, it's often called UCaaS (Unified Communications as a Service).

SIP trunking

A SIP trunk is a virtual phone line that connects a business's own phone system (PBX) or software to the phone network over the internet. Instead of renting physical lines, you buy trunks or channels (each channel handles one call at a time).

SIP trunking is popular with:

• Businesses that already own a PBX

• Call centers using dialer software

• Companies that want more control over their setup and lower costs at scale

Dedicated SIP trunks are reserved for your traffic only, so you get consistent capacity even when the network is busy.

Hosted dialers and call center platforms

Call centers often use dialing software that calls numbers automatically and connects answered calls to available agents:

• Preview dialer: the agent sees the customer's details and chooses when to dial.

• Power dialer: dials the next number automatically as soon as an agent is free.

• Predictive dialer: dials several numbers at once, predicting when agents will be free, so agents spend more time talking and less time waiting.

These dialers need dialer termination: voice routes and capacity designed specifically for high volumes of short, fast outbound calls.

Wholesale VoIP

Wholesale VoIP providers sell voice capacity in bulk to other carriers, VoIP resellers, call centers and large enterprises. Instead of phone features, they focus on routes, call quality, capacity and per-minute rates. See section 18 for a full explanation.

7. What do you need to use VoIP?

1. An internet connection

A reliable broadband connection: fiber, cable or a dedicated business line. For businesses, a stable connection matters more than raw speed. See section 8 for exact bandwidth numbers.

2. A VoIP provider

The company that connects your calls to the rest of the world, gives you phone numbers and runs (or connects to) your phone system.

3. A device to make calls

Any of these work:

• IP phone (desk phone): looks like a normal office phone but plugs into your network instead of a phone line.

• Softphone: an app on your computer or mobile that works like a phone. Use it with a headset for the best quality.

• ATA (Analog Telephone Adapter): a small box that lets you plug an ordinary phone into your internet router.

• Web browser: many systems now let you call directly from a browser tab using WebRTC.

4. The right network equipment

• A router that supports QoS (Quality of Service), which lets you give voice traffic priority over downloads and video streaming

• Optionally, a separate network (VLAN) for phones in larger offices

• Good headsets for agents, ideally with noise cancellation

8. How much internet speed does VoIP need?

Each call uses a small, steady amount of bandwidth in both directions (upload and download). The exact amount depends on the codec, plus some extra "overhead" for the packet labels.

How to calculate what you need: Multiply the bandwidth per call by the maximum number of calls you'll have at the same time (called concurrent calls), then add a buffer of about 20–30%.

Example: A call center with 50 agents on G.711 calls at the same time needs about 50 × 100 kbps = 5 Mbps upload and 5 Mbps download just for voice, plus room for everything else on the network.

Upload speed matters. Many internet plans have much lower upload than download speeds. Since VoIP sends your voice upward, always check your upload speed.

9. What affects VoIP call quality?

When people complain about VoIP, it's almost always about one of these issues. Here's what each one means and how to fix it.

Latency (delay)

The time it takes for your voice to reach the other person. The ITU (International Telecommunication Union) recommends keeping one-way delay under 150 milliseconds for natural conversation. Above that, people start talking over each other.

Causes: long or indirect routes, slow connections, too many hand-offs between carriers.

Jitter

Variation in how long packets take to arrive. Some arrive quickly, some late. High jitter makes voices sound choppy or robotic. Under 30 milliseconds is generally considered good.

Causes: busy networks, Wi-Fi interference, no QoS.

Packet loss

When some packets never arrive. Small amounts are hidden by the codec, but more than about 1% becomes noticeable as gaps, clipping or garbled words.

Causes: overloaded connections, faulty equipment, poor-quality routes.

Echo

Hearing your own voice back. Usually caused by speakerphones, cheap headsets or poorly configured equipment.

One-way audio

Only one person can hear the other. This is usually caused by a firewall or router setting (often related to NAT, the way routers share one public IP address among many devices) blocking the audio packets. It's a configuration issue, not a fault with VoIP itself.

Dropped calls

Calls cut off mid-conversation. Common causes are an unstable internet connection, firewall timeouts or poor-quality routes.

How quality is measured: MOS

MOS (Mean Opinion Score) rates call quality from 1 (bad) to 5 (excellent). A score of 4.0 or above is considered good business quality. G.711 calls on a clean network typically score around 4.1–4.4.

Route quality: the part you can't fix yourself

Even with perfect office equipment, a call can still sound bad if the provider sends it over poor-quality routes. Some low-cost providers pass calls through several cheaper carriers in a chain, and each hand-off adds delay and risk. That's why route quality is one of the most important things to ask a provider about.

Quick checklist to improve call quality

• Use a wired (Ethernet) connection instead of Wi-Fi for desk phones and agents

• Turn on QoS on your router and prioritize voice traffic

• Make sure you have enough upload bandwidth for your peak number of calls

• Use good headsets

• Choose a provider with high-quality, direct routes and active network monitoring

10. Phone numbers on VoIP: DIDs, toll-free and porting

DID numbers

A DID (Direct Inward Dialing) number is a phone number that rings directly to a person, team or system on your VoIP setup. VoIP numbers are not tied to a location: a business in one city can have local numbers in many others, so customers see a familiar local area code.

In the US and Canada, phone numbers follow the NPA-NXX-XXXX format:

• NPA = the area code (for example, 863)

• NXX = the exchange (the next three digits)

• XXXX = the line number

Wholesale providers often price calls by NPA-NXX, because the cost of reaching a number depends on which carrier and area it belongs to.

Toll-free numbers

Numbers such as 800, 888, 877 and 866 in the US and Canada. The business pays for the call instead of the caller.

Number porting (keeping your number)

You can usually move your existing phone numbers to a VoIP provider. This is called number porting, or LNP (Local Number Portability). You'll typically need:

• A recent bill from your current provider

• The account number and PIN or passcode

• The authorized account holder's name and service address

• A signed authorization form (often called an LOA, Letter of Authorization)

Porting usually takes from a few days to a few weeks. Don't cancel your old service until the port is complete, or you could lose the number.

Caller ID name (CNAM)

CNAM is the business name that shows up on some phones when you call. Your provider can usually set or update it for your numbers.

11. VoIP features

Business VoIP systems usually include features that cost extra, or weren't available at all, on traditional lines:

Call handling

• Auto attendant / IVR (Interactive Voice Response): "Press 1 for sales, 2 for support"

• Call forwarding, transfer, hold and park

• Ring groups and hunt groups: ring several people at once or one after another

• Call queues with hold music and position announcements

• ACD (Automatic Call Distribution): sends incoming calls to the right available agent

Productivity

• Voicemail-to-email and voicemail transcription

• Find-me/follow-me: calls ring your desk, then your mobile

• Mobile and desktop apps using the same business number

• Conference calling

Management and data

• Call recording for training, quality and compliance

• Real-time dashboards and call analytics

• CDRs (Call Detail Records) for every call

• CRM integration: see customer details when the phone rings and log calls automatically

Call center tools

• Predictive, power and preview dialers

• Live call monitoring, whisper (coach an agent without the customer hearing) and barge (join the call)

• Local presence dialing: show a local number to the person you're calling

12. Advantages of VoIP

• Lower costs: cheaper long-distance and international calls, no separate phone lines to rent, and lower costs at high volume.

• Scalability: add or remove lines in minutes, which is ideal for growing teams, seasonal spikes and new campaigns.

• Flexibility: work from the office, home or anywhere with internet, using the same business number.

• Advanced features: IVR, recording, analytics and integrations included or available as add-ons.

• Global reach: local numbers in many cities and countries, and affordable international calling.

• Better voice quality: with a good connection and good routes, VoIP can sound as good as or better than a traditional line, and HD voice codecs can sound better.

• Easier management: control everything from one online dashboard.

• Business continuity: calls can be rerouted to mobiles or another office if one location goes down.

13. Disadvantages of VoIP (and how to solve them)

14. VoIP security and fraud

Because VoIP runs over networks, it needs the same security care as any other business system.

Common threats

• Toll fraud: criminals break into a phone system and use it to make large numbers of expensive calls, often overnight or over a weekend. The business gets the bill.

• IRSF (International Revenue Share Fraud): a type of toll fraud where criminals send calls to expensive international numbers they profit from.

• Eavesdropping: intercepting unencrypted calls.

• Caller ID spoofing: faking the number that appears on the receiving phone.

• DoS (Denial of Service) attacks: flooding a system with traffic so real calls can't get through.

• Account takeover: stolen login details used to change settings or forward calls.

How to protect your business

• Use strong, unique passwords for every phone, extension and admin account, and change default passwords immediately

• Turn on two-factor authentication for admin portals

• Use encryption: TLS for SIP signaling and SRTP for audio

• Restrict international calling to the countries you actually need

• Set spending limits and alerts for unusual activity

• Use IP allowlisting so only your approved systems can send calls

• Keep phone and PBX software updated

• Choose a provider with 24/7 fraud monitoring that can detect and block suspicious traffic automatically

15. STIR/SHAKEN, robocalls and caller ID

Robocalls and spoofed numbers have made people wary of answering unknown calls. STIR/SHAKEN is the framework introduced to fight this in the US and Canada.

What it is: a system that digitally "signs" calls so the receiving network can verify that the caller ID hasn't been faked. (STIR and SHAKEN are the names of the technical standards: Secure Telephone Identity Revisited, and Signature-based Handling of Asserted Information using toKENs.)

How it works: when a call starts, the originating provider gives it an attestation level based on how well it knows the caller and the number:

• A (Full attestation): the provider knows the customer and confirms they're allowed to use that number.

• B (Partial attestation): the provider knows the customer but can't confirm they're allowed to use that number.

• C (Gateway attestation): the provider is passing the call along but can't verify where it came from.

The receiving carrier uses this information, along with its own analytics, to decide whether to show the call as verified, label it as possible spam or block it.

Why it matters for businesses: if your calls are poorly attested or your numbers get flagged, more of your calls show as "Spam Likely" or go unanswered. For call centers, this has a direct impact on answer rates and revenue.

How to protect your answer rates:

• Use a provider that fully supports STIR/SHAKEN

• Make sure your numbers are registered to your business with your provider

• Don't make too many calls from a single number, and avoid patterns that look like spam

• Monitor your numbers' reputation and replace any that get flagged

• Follow calling rules and do-not-call regulations in the regions you call

16. Emergency calls (911) on VoIP

Because VoIP isn't tied to one physical location, emergency calls need extra setup.

• E911 (Enhanced 911): in the US, interconnected VoIP providers must support 911 calls and send the caller's registered location to emergency services. Canada has similar 911 requirements.

• Register your address: every number or phone must have the correct registered address. If staff work from home or move between offices, update their addresses.

• Kari's Law (US): multi-line phone systems must let users dial 911 directly, without a prefix like "9," and must notify someone on site (such as reception or security) when 911 is called.

• RAY BAUM'S Act (US): 911 calls must send a "dispatchable location," meaning detailed enough (like a floor or suite number) for responders to find the caller.

• Power and internet outages: VoIP phones can't make any calls, including 911, without power and internet. Keep a mobile phone available as a backup.

17. How VoIP pricing works

VoIP pricing depends on the type of service.

Cloud phone systems (hosted PBX / UCaaS)

Usually a monthly fee per user, often including a calling allowance (or unlimited calling within certain countries) and a set of features. Higher tiers add more features.

SIP trunking

Usually priced by:

• Per channel: a monthly fee for each concurrent call, often with unlimited inbound or local calling

• Metered: pay per minute for the calls you make

• Flat-rate plans: a fixed monthly price for a set amount of usage or capacity

Wholesale and call center voice

Usually per minute, with rates that vary by destination. Watch out for:

• Billing increments: how calls are rounded. 1/1 means billed per second. 6/6 means a 6-second minimum, then 6-second blocks. 60/60 means every call is rounded up to a full minute. For high volumes of short calls, this makes a big difference to the bill.

• NPA-NXX (rate deck) pricing: in the US and Canada, the rate can vary by area code and exchange.

• Interstate vs. intrastate rates: in the US, calls within a state can be priced differently from calls between states.

• Minimum commitments and setup fees

Other costs to check

• Phone numbers (DIDs) and toll-free numbers, monthly and per minute

• Number porting fees

• Equipment (IP phones, headsets, ATAs)

• Regulatory fees and taxes

• Hidden fees: always ask for a full breakdown before signing

The real cost of cheap VoIP

The lowest per-minute rate isn't always the cheapest option. Dropped calls, poor audio and low answer rates cost businesses far more in missed sales, wasted agent time and damaged reputation than they save on the phone bill.

18. Wholesale VoIP explained

Wholesale VoIP is the layer of the industry most people never see. It's how voice traffic travels between carriers around the world.

Who uses wholesale VoIP?

• Carriers and telecom operators that need routes to destinations they don't reach directly

• VoIP resellers and service providers that sell voice services under their own brand

• Call centers and BPOs sending large volumes of calls

• Large enterprises with heavy calling needs

Key terms

• Termination: delivering a call to its destination network (outbound).

• Origination: receiving calls from the phone network onto VoIP (inbound), usually through DIDs.

• Voice routes: the paths calls take between networks to reach their destination.

• Tier-1 routes: high-quality routes through major carriers with direct interconnections, with fewer hand-offs. This means less delay, better audio and fewer failed calls.

• CLI / non-CLI routes: CLI (Calling Line Identification) routes deliver your caller ID correctly to the person you're calling. Non-CLI routes may not, which reduces answer rates and trust.

• Grey routes: unofficial routes that bypass normal interconnection agreements to cut costs. They're cheaper but less reliable, and can cause quality and compliance problems.

• A-Z termination: coverage to all (or nearly all) destinations worldwide.

• CPS (Calls Per Second): how many new calls you can start each second. Critical for dialers.

• Concurrent calls/channels: how many calls can be active at the same time.

• Interconnect: the technical link between two carriers' networks.

How wholesale call quality is measured

• ASR (Answer-Seizure Ratio): the percentage of call attempts that are answered. A higher ASR generally means better routes, though it also depends on the type of traffic.

• ACD (Average Call Duration): the average length of connected calls. Very short averages can signal quality problems or the wrong type of traffic for the route.

• PDD (Post-Dial Delay): the time between dialing and hearing ringing. Shorter is better.

• NER (Network Effectiveness Ratio): measures how well the network delivers calls, regardless of whether the person answers.

19. How to choose a VoIP provider

Use these questions to compare providers, whether you need a phone system for a small team or high-volume voice for a call center.

1. What type of service do you need?

• Small or mid-sized office → hosted PBX / cloud phone system

• Own PBX or dialer software → SIP trunking

• High-volume outbound calling → dialer termination

• Carrier, reseller or very large volumes → wholesale VoIP

2. How reliable is the network?

• What uptime do they commit to, and what happens if they miss it?

• Which routes will your calls use? Are they Tier-1, direct routes?

• Do they monitor call quality and the network 24/7?

3. Can they handle your volume?

• How many concurrent calls and how much CPS can they support today?

• How quickly can they increase capacity?

• Are dedicated trunks available?

4. Will your calls get answered?

• Do they fully support STIR/SHAKEN?

• Do they deliver your caller ID correctly (CLI routes)?

• Can they help protect your numbers' reputation?

5. What support do they offer?

• Is technical support available when you need it, including nights and weekends?

• Will you speak to voice specialists or a general help desk?

• Is there a clear way to escalate urgent issues?

6. Is the pricing clear?

• Are the rates transparent, with no hidden fees?

• What billing increments do they use?

• Will they give you a custom quote based on your actual traffic?

7. Is it secure?

• Do they offer encryption, IP allowlisting, and fraud monitoring?

• Can you set international restrictions and spending limits?

8. Can you test before committing?

• Will they let you run test calls or a trial on your real traffic?

• Can they share quality reports (ASR, ACD, PDD) during the trial?

20. Frequently asked questions

• Is VoIP the same as the internet? No. VoIP is a way of sending voice calls over the internet or private IP networks. It uses your internet connection the same way email or video streaming does.

• Can I call regular phones and mobiles with VoIP? Yes. Your VoIP provider connects your call to the regular phone network, so you can call any landline or mobile number.

• Do I need a special phone? No. You can use an IP phone, a softphone app on your computer or mobile, a web browser, or a regular phone with an adapter (ATA).

• Can I keep my current phone number? In most cases, yes, through number porting. Don't cancel your old service until the port is complete.

• Does VoIP work during a power outage? Not without backup. Use a battery backup (UPS) for your router and phones, and set up call forwarding to mobiles.

• Is VoIP secure? It can be very secure when set up properly: strong passwords, encryption (TLS and SRTP), international call restrictions and a provider with fraud monitoring.

• Is VoIP call quality as good as a landline? Yes. With a stable connection, a good codec and high-quality routes, VoIP sounds as good as a landline, and HD voice can sound better. Poor quality almost always comes down to the network or the provider's routes.

• How many calls can I make at once? That depends on your bandwidth and your plan. Each G.711 call needs about 100 kbps in each direction. Your provider also sets limits on concurrent calls and CPS.

• Is VoIP legal? Yes, in the US, Canada, and most countries. A few countries restrict some VoIP services, so check local rules if you operate internationally.

• What's the difference between VoIP and SIP? VoIP is the overall technology of making calls over IP networks. SIP is one of the protocols VoIP uses to set up and control those calls.

• What's the difference between VoIP and a SIP trunk? VoIP is the technology. A SIP trunk is a specific VoIP service that connects your own phone system or dialer to the phone network.

• Why do my VoIP calls sound choppy? Usually jitter or packet loss, caused by a busy network, Wi-Fi, not enough upload bandwidth or poor routes. Use a wired connection, turn on QoS, and check your bandwidth. If it continues, ask your provider about route quality.

• Why can only one person hear on a call? That's one-way audio, usually caused by firewall or router (NAT) settings blocking the audio. Your provider or IT team can fix it with the right configuration.

• Can I use VoIP for a call center? Yes. Most call centers run on VoIP. Make sure your provider offers enough CPS and concurrent calls, dialer termination, STIR/SHAKEN support, and high-quality routes.

Ready to get started with business-grade VoIP?

Dial Edge Telecom provides carrier-grade voice infrastructure for call centers, carriers, and enterprises across the US and Canada:

• Tier-1 voice routes for crystal-clear, consistent call quality

• 99.99% uptime

• Wholesale voice termination

• Dedicated SIP trunks

• Premium dialer termination

• DID numbers and hosted dialers

• Transparent rates. No hidden fees.

Reliable voice for the US and Canada. Enterprise performance.

Get a custom VoIP rate quote, email sales@dialedgetelecom.com or call +1-863-513-5237.