6 Best SIP Trunking Providers of 2022
The success of SIP Trunking in improving the business communications of SMBs, SMEs, and Large-scale businesses and enterprises has made the industry worth over $10 billion today.
However, with a forecasted CAGR growth of 10.7% in coming years and few limitations as to the standard of service, more subpar service providers are expected to trod into the market seeking to earn a profit.
This presents a problem for consumers like you who are looking for quality SIP trunking solutions amidst pressing needs to upgrade communication systems.
With this article, you have all the information you need about the best SIP Trunking providers in the market today and some extra that could help you understand what the whole concept is all about.
Best SIP Trunking Providers
Best SIP Trunking Service With Extensive Features and Integrations
Twilio is a business communication solution provider founded on the 13th of March 2008 by Jeff Lawson, Evan Cooke, and John Wolthuis to provide businesses with mediums to communicate virtually, through VoIP.
One of Twilio's multiple VoIP services is its SIP Trunking solution that gives you a complete set of features to achieve a seamlessly scalable business communication system.
With Twilio's Elastic SIP Trunking, you enjoy agility in developing your business communication systems through self-service options and innovative cloud features, all coming along with unlimited capacity in holding your SIP channels.
Within minutes, you can achieve global call coverage to at least 100 countries through its console or the API provided to you.
You get to operate international business communications activities without worrying about their quality or the overall efficiency of business communications.
Comprehensive security and fraud prevention features, alongside an expansive regional inventory and disaster recovery capabilities, are also made available, giving you all the security and reliability needed to run your business communications.
Inbound and outbound PSTN connectivity can be integrated into your contact center by moving it to the cloud, allowing you to guarantee high availability to clients and customers.
While upgrading your phone system to a PSTN powered by VoIP, you also have the option of keeping your current carrier through its BYOC (Bring Your Own Carrier) support.
Mass notifications for quick campaign voice and text messaging as well as sales and market automation for marketing outreach or political campaigns are also made available to you.
Twilio's Elastic SIP Trunking can also serve as a backup or overflow solution, giving you a chance to go on with business communications when your current SIP provider is down.
What more? This backup service won't cost you money unless you use it.
Twilio's pricing is shaped around a Pay-As-You-Go scheme, meaning you only pay for what you use and no extra.
You can also get charged based on a committed-use pricing scheme.
The overall cost is determined by the number of calls per second preset by you.
Very Efficient SIP Trunk Provider For North American Call Coverage
Founded in 2006, Nextiva aims to provide businesses with Unified communication solutions that help to comprehensively facilitate business communications between employees and customers.
Nextiva's SIP Trunking services offer you a chance to seamlessly transition your legacy PBX system to one powered by cloud-based technology.
With Nextiva, you have access to a supervisor dashboard that allows you to monitor call flows in real-time, giving you a chance to ensure that you always have sufficient agent coverage on inbound calls.
You can say goodbye to expensive hardware equipment and save up to 70% on your current PBX billing through cost-effective calling solutions.
Enjoy unlimited calls across Canada, The United States, and Puerto Rico as well as reduced international call rates when compared to traditional phone systems.
Offering unified communications solutions, Nextiva allows you to connect a hosted PBX to your SIP trunk via a PSTN, giving you a completely cloud-based phone system.
Through adapters designed especially for your traditional business phone system, you also get to keep your existing hardware and avoid the costs of purchasing new equipment.
Dedicated support to oversee migration and porting, as well as network analysis and troubleshooting, are also made available to you.
To cap it off, Nextiva promises you a 99.99% uptime, 8 points of presence (PoPs) across international locations, access control to mitigate fraud, and E911 coverage in the US and Canada.
You are charged according to two different pricing plans: Metered and Unmetered.
The metered plan gives you low-cost call rates while the unmetered plan offers you unlimited calls to the US, Canada, and Puerto Rico.
SIP Trunking Service With Extended VoIP Options For Collaborative Team Operations
RingCentral was founded in 1999 by Vlad Schmunis and Vlad Vendrow and focuses mainly on providing businesses with Unified communications solutions, SIP Trunking being a minor part of its service provision.
Through SIP Trunking, RingCentral allows you to integrate your third-party IP phone systems with its advanced VoIP services, enabling you to make and receive calls using its servers.
A host of team collaboration options are provided alongside VoIP services, presenting you and your team with one of the best video conferencing software in the market today.
Coupled with these high-quality video conferencing options are high carrier-grade calls that allow your business to be readily available to over 40 countries worldwide.
The much-needed security and protection required by your communication system are also made available by RingCentral through conversation and data encryption.
You get to keep your employees and customers safe from hackers and feel comfortable sharing confidential information without any fear of it being stolen.
Intelligent inbound call routing options, as well as powerful outbound call and campaign options, are also provided by RingCentral to ease vast customer reach.
RingCentral charges universal fees for its SIP and VoIP services and offers you high-quality VoIP desk phones to utilize and integrate its SIP protocols into your cloud phone system.
Prices are mainly determined by the number of users expected to use the VoIP services.
Offers Highly Secured, Cross-Continental Inbound And Outbound Call Options
Plivo was founded in 2012 by Venky Balasubramanian and Michael Ricordeau and was originally intended to be an open-source telephone project.
Today, Plivo serves as an enterprise-grade communications platform that offers premium carrier network services alongside voice and messaging solutions with connectivity to over 190 countries.
Nicknamed “Zentrunk“, Plivo presents you with quality SIP trunking services built and optimized for existing network infrastructure, allowing you to interoperate with standard soft switches and IP PBXs.
With a global infrastructure that consists of 6 points of presence (PoPs) across five continents, you are assured of low latency and high-quality international voice calls.
International outbound calls can be made seamlessly and you can receive inbound calls from over 70 different countries; these capabilities are accompanied by local, national, and toll-free number options.
Multi-level security options through IP and Credential-based authentication as well as TLS and SRTP trunk encryption are also made available by Plivo.
You can protect your SIP trunk and communication system from unauthorized access.
Automated fraud detection and alerts are also presented to you and your communication system is promised a 99.99% uptime.
Plivo's extensive SIP services are rounded off by a self-service portal that allows you to personally manage SIP trunks, accounts and have access to analytics data and logs.
Plivo's Zentrunk pricing is based on a Pay-As-You-Go scheme, with prices being determined by your region of operation and the type of number you intend to operate with.
You also enjoy volume discounts as you scale up your communication system.
Cheap SIP Trunking Service With the Widest Call Coverage
Vonage was founded in 2002 to provide unified communications solutions in dealing with your business communications needs.
Vonage SIP Trunking makes it easy for you to seamlessly expand the capabilities of your traditional PBX system and give it global coverage within minutes.
It connects your company to individuals in 225 countries and territories worldwide through over 1600 telecommunication networks.
Through its dashboard, you have access to useful insights that can be put to good use and you also have a choice to create your personalized dashboard through its management API.
You can link your PBX system to the cloud and achieve faster communications while saving costs and avoiding traditional capacity limitations.
Your contact center is efficiently scaled up and you can maintain quality global inbound and outbound calls from anywhere and at any time.
Easy-to-use developer tools are also made available, alongside automatic location-based routing that helps to reduce latency and maintain high voice quality on calls.
Vonage protects your communication system through TLS encryption and an IP authentication option.
You get to add a layer of security over your VoIP infrastructure and protect your privacy and data integrity.
Multi-codec options as also provided and enable you to dictate the sound quality of your phone system according to bandwidth and computational specifications.
Number porting options that allow your customers to continue using old numbers and help you to avoid purchasing new numbers are also made available.
Vonage also makes use of an elastic Pay-As-You-Go pricing scheme based on call rates per minute.
Best SIP Trunk For Very Large Enterprises And Organizations
Bandwidth offers your business high-quality SIP Trunking services that give you and customers reliable voice calling options as well as flexible usage-based pricing and software tools.
With Bandwidth’s cloud-based native carrier network, high-quality voice calling is never a problem and you get to eliminate costly and inflexible PRI lines from your communication system.
An intuitive dashboard and API suite present you with software-driven number management options.
These number management options make it easy for you to order phone numbers, manage ports, and get real-time insights into telecom cost centers.
Better workstation management is achieved through these and your communication system remains optimized to your taste.
Advanced E911 capabilities that maintain compliance with 911 regulations can also be integrated into your PBX and business phone infrastructure.
You also have a BYOC (Bring Your Own Carrier) option, allowing you to choose the carrier you want to be integrated into your contact center, conferencing, and unified communication system.
Bandwidth also provides you with dedicated specialists that help you through migration, scaling, and changes in telecommunications requirements.
Alongside all these advanced SIP Trunking features, Bandwidth also makes other intuitive options available for your communication needs. These include domestic and International/long-distance calling, SMS capability, toll-free calling, and real-time disaster recovery and IP failover control.
Bandwidth makes use of a Pay-As-You-Go scheme for its pricing, with different rates charged for voice calls, video calls, messaging, and phone numbers.
You can also request a custom pricing package if you operate a business with high-volume or enterprise-level calls.
What is SIP Trunking?
Before knowing what SIP Trunking is, you must know what the terms “SIP” and Trunking mean.
An abbreviation for Session Initiation Protocol, SIP is an internet protocol that runs on VoIP technology and allows you to enjoy voice and other Unified communications services over the internet.
SIP serves as an alternative for Primary Rate Interface (PRI) as, rather than making use of analog lines, calls are transmitted over the internet through SIP channels.
SIP serves as a solution that helps you integrate VoIP abilities into your physical or on-site PBX system and get rid of the complexities that come with the use of physical analog lines.
Where a PBX system is not compatible with SIP, adapters can be made available by a provider to give your regular PBX proper VoIP capabilities.
Trunking on the other hand means the grouping or combination of SIP channels.
A SIP Trunk serves as a backbone that can hold an unlimited number of SIP channels.
This translates to you having unlimited telephone lines on one telephone system without having to purchase extra equipment to host more telephone lines.
SIP is usually confused with VoIP as they are very much similar, however, it is only one of the protocols enabled by VoIP.
How Does SIP Trunking Work?
SIP trunking is one of the most popular and relieving business communications solutions available in the world today.
However, a lot of equipment and protocols come together for it to work effectively.
The basic equipment and components needed for SIP trunking include:
- Internet connection
- IP PBX/ SIP compatible PBX
- VoIP phones or VoIP adapters for integration into traditional PBX that are not compatible with SIP
- VoIP app for sending out calls through phones or other devices.
SIP is an application layer protocol, meaning it involves multiple protocols on the internet that work together to make VoIP calls feasible.
If and when you make the required equipment and components available for your business. Here is how SIP trunking works.
1. Voice/Audio Data Is Encoded With Codecs
Before being sent through SIP channels, voice or multimedia data is converted into data packets and encoded with codecs to be sent through the internet.
Codecs are very important in VoIP and SIP Trunking, and there are two popular codecs used by providers to facilitate internet calls: The G.711 codec and The G729 codec.
The G.711 codec is used for transmitting uncompressed, high-quality calls and is the standard codec used in SIP.
This means that with the G.711 codec, you get the best voice or multimedia quality transmitted, however, a powerful bandwidth is required between the caller and receiver.
SIP providers can help to create a strong connection between callers that are geographically far apart and provide great call quality as if they were in the same room.
The G729 codec is used for compressed voice data and works on calls operating through a limited bandwidth or where a dedicated internet connection is unavailable.
If you or your callers operate on ADSL or FTTC connections to make VoIP calls, the G729 codec comes in place to compress the data and make calls achievable but in low quality.
2. Session Description Protocol Defines The Data Type
To make sure that the whole SIP call is possible, SDP defines the data sent by a caller, compares the devices of both the caller and receiver, and determines if the receiver can support the data type.
3. Makes use of Real-time Transport Protocol (RTP)
To facilitate real-time exchange of voice or multimedia data between the caller and receiver, SIP works with the Real-time Transport Protocol.
The relationship between RTP and SIP is managed by an RTP Control Protocol which keeps the exchange of data between the caller and receiver high-quality and accurate.
4. RTP packets are transferred through TCP or UDP
Encoded audio data is then transported alongside RTP packets through transport layers.
The Transmission Control Protocol and the User Datagram Protocol are the transport layers that help to facilitate SIP calls and the User Datagram protocol acts as the best transport layer for delivering VoIP calls through SIP channels.
5. Codec Converts Back To Original Form
When the audio data packets reach the destination, either at the caller's endpoint or receiver's endpoint, the codec then converts it back to the original form for replay.
Advantages Of SIP Trunking
SIP and VoIP have come to stay and the use of traditional phone systems and PRIs is fast disappearing from the business world as the years go by.
SIP holds a lot of advantages over Primary Rate Interface (PRI) and some of these include;
1. Reduced and Scalable Cost
Arguably, the most recognized and cherished advantage of using SIP trunking in place of a Primary Rate Interface (PRI) is the amount of cost that can be saved through it.
A PRI makes use of analog lines, meaning that a lot of technical support is needed to make sure that telephone lines are in good shape.
With SIP trunking, you get to pay no maintenance or IT fees to external experts or dedicated employees to take care of analog lines or solve highly technical problems.
You also avoid high monthly costs through very low call rates after paying a monthly subscription.
Some SIP providers even grant you unlimited call rates included as part of the subscription.
SIP Trunking also gives you a chance to properly scale costs without having to pay for extra components that are not needed.
One PRI has 24 channels. Meaning you have no choice but to pay for extra 23 channels or telephone lines that are not needed for your business.
With SIP Trunking, you can purchase only one SIP channel, which translates to more scalable and careful cost servicing as your communication needs grow.
What more? The extra cost can be saved if you operate a traditional PBX. SIP adapters can be used to give traditional PBX VoIP capabilities without needing to completely change to an IP or SIP compatible PBX.
A PRI requires physical desk phones to work.
The need for desk phones significantly limits the way you receive quality business calls outside the office.
SIP Trunking requires only internet-capable devices and VoIP apps installed on them, giving you a chance to receive business calls from anywhere and at any time.
You have access to dynamic communication options which make you available to your customers and partners even at the most informal and remote of locations.
According to Forbes, 92% of consumers will avoid dealing with a company after two or three bad experiences with it.
Your communications system needs to be up at all times to guarantee that customers or partners can reach you without being faced with hurdles.
SIP Trunking makes use of the internet and one thing is sure: the internet will always be up and available for access no matter the situation.
SIP providers also have backup servers at different locations in case of unfortunate events, presenting you with highly reliable options that help you remain available for both customers and partners at all times.
4. Extensive Features
SIP Trunking offers more than just seamlessly and cost-effectively transmitting calls from one endpoint to another.
You enjoy access to a lot of advanced call management options at no extra cost.
Features like advanced call forwarding, caller ID, email optimization, and advanced voicemail management, among others, are made available to you.
Call analytics and reports can also be generated when using SIP, providing you with a chance to pinpoint problems and easily make decisions on how to improve business communications.
How Can SIP Trunking Improve Business Communications
A lot of businesses have welcomed this shift in preference and your business communications shouldn't stay limited by analog telephone lines and the long list of complexities that come with them.
SIP Trunking offers a lot of perks to keep your business communications very much intact and efficient and some of these include;
1. You Save Money On Equipment And Calls
A business venture operates with the sole aim of making as much profit as possible. Saving as much money as you can is one way of increasing your ROI and SIP Trunking offers you the communications solutions needed to achieve this.
Your business has access to relatively low call rates and even unlimited call rates from specific providers.
You can also switch from a traditional phone system to SIP Trunking with the use of adapters, helping you avoid paying for new equipment.
One perk intertwined with the low costs of SIP Trunking is the scalability it provides to your business communication needs.
As mentioned earlier, One PRI comes with 24 channels or telephone lines. When there is a need to scale up your communication system, you have to pay an expensive fee for extra telephone lines that are not needed.
SIP Trunking offers the scalability required to properly streamline business needs with your expenditure.
SIP Trunking offers your business reliable solutions that keep your communication system up and keeps you available at all times.
Working with the internet, service outages are a thing of the past, and where a connection can be established, communication can be established.
SIP providers also operate backup servers which make sure that in case of a failed server or unfortunate event, your communication system remains online to take care of calls.
4. Flexible and Portable Communication
Remote working has become one of the most definitive parts of the business world today, more so after the advent of the Covid-19.
Do you want your business to be limited only to the office or any location where a desk phone is situated? Definitely not.
SIP Trunking offers you a chance to relate with your customers and business partners from anywhere and at any time.
Business communications are never kept on hold.
How to Choose The Right SIP Trunk Solution
There are a lot of SIP providers in the market today and picking the best can be a tough task.
Even the most highly rated SIP Trunk providers may be incompatible with your business communication needs.
There are a lot of criteria involved in picking the right provider for your business.
1. Acknowledge Your Business Communications Needs
Recognizing how your business communication is shaped and the different components and features needed to improve it is the first step in picking the best SIP trunk provider.
Whether small, medium or large scale, every business has communication needs that are peculiar to it and important in achieving business goals.
Do you need Local numbers only? Or does your communication spread across countries and wide geographical areas? How many calls does your business receive? How many employees do you manage in your business? Do you even need SIP and other VoIP unified communications solutions? These are a few of the important questions that need to be asked before committing your communication system into the hand of a SIP provider.
2. Take Note Of A Provider's Coverage
Communication is the exchange of information from one endpoint to another.
The type of business you run determines if you need local or international numbers or both.
SIP providers facilitate communication between defined geographical locations and you must select one that operates in your locations of interest.
Look for a provider with at least 1 point of presence (PoP) in every possible region of operation.
If you run a cross-border business venture, consider a SIP provider that can facilitate a wide range of global inbound and outbound calls.
A SIP provider with international call solutions keeps your global communication needs well covered and efficiently managed.
You get to enjoy cheaper voice calls due to the absence of a middleman and you also achieve higher quality voice calls due to dedicated international servers.
3. Consider Its Overall Cost
One of the advantages of SIP Trunking is cost-saving and this is one thing you don't want to miss out on.
The PAYG structure of purchasing SIP channels is always greater than the upfront payment for PRI channels.
Different providers charge different prices for SIP Trunking and you must choose one that offers great value for money.
A cheap SIP Trunk provider may not be the best for you. This is because the quality of service may be sacrificed for the cost of service and, very possibly, your business communication needs are not entirely covered by the features offered.
Consider the SIP provider with the best balance between quality of service and cost as relating to your business communication needs.
4. Ensure It Offers Great Voice Quality
Why settle for a SIP provider known to offer low-quality voice calls?
If high-quality communications aren't achievable, the SIP is useless for your business and this could be detrimental to the satisfaction of partners, employees, and customers in relating with each other.
Looking for and selecting a SIP provider that works with Tier-1 voice carriers is one step in achieving great voice quality. It is advised that you avoid SIP providers that use Least Cost Routing (LCR).
LCR compromises quality for cost and this is not a good thing to be associated with your business communications.
Another point to note is that some SIP providers tend to rebrand the SIP services of another provider and only serve as middlemen in the whole operation.
These rebranded SIP services tend to be cheaper than other SIP providers, however, the quality of services is always sacrificed.
5. Security and Reliability Are Important
Security is an increasing concern for businesses, especially how the internet has shaped out to be in recent years.
Hackers and other fraudulent individuals roam the internet and you don't want to be that one person that falls victim to them.
Important information such as passwords and confidential pins and codes, among others, is regularly exchanged between your business and employees, partners, or customers.
SIP providers present multiple solutions in dealing with the security issues involved in safeguarding business communications.
IP authentication is one of these and helps to keep intruders out of your communication databases.
Trunk encryption through Transport Layer Security (TLS) and Secure Real-Time Transport Protocol (SRTP) also helps to keep the data transmitted through SIP channels safe and secure.
The information remains useless even when there is access from an intruder.
Check out the existing security architecture of a SIP Trunk provider and ensure that what it offers is sufficient in safeguarding your business communications.
The customer servicing, uptime rating, and backup plans of a SIP provider are also important factors to consider before making a choice.
6. Look For Integrations With Existing Equipment
You don't want to discard existing infrastructure and waste time and money on setting up new equipment.
Look for a SIP provider that offers interoperability with your existing equipment and adapters for your traditional PBX system.
Number porting capabilities are also an important factor to look out for in ensuring that important existing components are kept.
How to Get Started with a SIP Trunking Service
Before getting started with SIP trunking, you need to determine the resources needed to run it effectively. Some of these include;
- Router and firewall that supports port address translations and static port mapping
- A domain name with a resolution to a static external IP
- The bandwidth of at least 30-120kbps for proper and achievable VoIP call handling.
- Existing and supporting VoIP provider.
After all these foundational components have been put in place, the next steps include;
1. Setting Up Your Account With A SIP-Compatible VoIP Provider
Not all VoIP providers offer SIP Trunking services and this should be acknowledged before proceeding to choose a service provider.
Some of the popular and highly rated VoIP providers that offer you quality SIP trunking services include:
After selecting one, create an account with it and select your preferred pricing plan as relating to your business needs
2. Test Your Firewall
A firewall serves as the middle man between your private PBX system and the public SIP channels running through the internet.
Test out your firewall to ensure that it is configured correctly, compatible with your SIP and audio ports, and is sufficient enough in keeping your communication system safe and secured.
3. Integrate Your VoIP Account Into Your System
First, select the options appropriate for your geographical location and the numbers that are assigned to your new SIP trunk.
Create a name for your VoIP account and compare the details for your hostname and outbound proxy with the details provided by your VoIP provider.
Assign the number of simultaneous calls that are allowed by your provider and set the authentication based on your registration details.
Specify how the routing on your main number should be set up (I.e how communications are run through SIP channels), and specify your DID numbers by adding the ones that are associated with your account.
Assign the number that you want to show up on your caller ID and save your settings.
4. Set Your Outbound Rules
Now you can create the outbound rules based on your communication infrastructure and needs and decide which kinds of calls will be routed over your SIP trunk.
You are allowed to set these rules according to the group or individual that is making the call, the length of the number, or the number that person is dialing.
You can also specify the elements that will trigger each rule and, for example, can create rules only for numbers that have a certain prefix, calls that come from a certain range of extensions, or calls from a selected extension group.
Making use of intuitive names that can be easily linked to the various departments and routes is great for easy call rerouting and management.
Specify how outbound calls will be routed and you can set alternative outbound rules that will step in when the first set of routes is not available.
After setting call handling rules, your communication system is ready to go and you can now enjoy all the perks and advantages that come with the use of SIP Trunks.
SIP Trunking Providers FAQ
Although SIP and VoIP are very closely related in both definitions and how they work, they are completely different and distinguishable.
VoIP defines all calls made over the internet. It covers every type of voice or video call facilitated by the internet.
SIP is only one of the protocols that makes use of VoIP to place calls over the internet.
While SIP needs VoIP, VoIP doesn't need SIP to work. This would be explained further.
SIP Trunking only involves the replacement of a Primary Rate Interface or analog lines with SIP channels that run through the internet.
An on-site PBX is still required to be integrated with a SIP Trunk and an existing VoIP service is required for it to go live.
You can use a Hosted or cloud PBX instead of a SIP trunk and this still involves VoIP, as long as communications are channeled through the Internet.
The cost of a SIP Trunking service is significantly determined by the cost of one SIP channel as charged by the provider chosen by you.
An average of $10 is usually charged for one SIP channel monthly on the lowest plans of SIP providers and an average of $25 monthly is charged on the highest plans that could offer you unlimited SIP channels.
Call rates can be unlimited but are usually charged for $0.001 per minute and other extra costs such as setup fees can also be charged by SIP providers.
Without an existing business phone system, making use of VoIP with a hosted PBX is a great choice when it comes to overall scalability, feature-set, and cost-effective business communications solutions.
However, if you have an existing on-site PBX system or legacy telephone lines, integrating SIP trunking will be one of the best moves for your business communications.
You get to keep your communication infrastructure and equipment while setting yourself up to save more cost on both inbound and outbound calls as well as maintenance.