SIP trunking 101: A complete guide to enhancing voice connectivity

SIP stands for Session Initiation Protocol. It is a communication protocol used for initiating, maintaining, modifying, and terminating real-time sessions involving video, voice, messaging, and other communications applications.

SIP trunking is a service provided by communication service providers. It uses the Session Initiation Protocol to connect a phone system to the public switched telephone network (PSTN) over the internet.  

SIP is used for call setup, management, and termination. It’s often offered as a replacement for traditional phone lines, known as Primary Rate Interfaces (PRIs), which use older technology.  

Many communication service providers around the world are adopting SIP trunking to replace their outdated services. By using existing network infrastructure like ethernet or Multiprotocol Label Switching (MPLS), they can reduce costs and simplify management. Plus, SIP trunking offers additional features compared to traditional phone lines. 

SIP trunking is great for businesses that need multiple phone lines. This includes everyone from small businesses to big companies with global sites. If you make a lot of calls or expect to make more in the future, it can really help your business. It offers a bunch of benefits that you’ll find useful. 

These are some of the most common benefits for businesses:

Phone lines can be expensive, especially if you have unused lines or frequently make international calls. You also have to pay for hardware and maintenance. With SIP trunking, you’re charged based on the number of concurrent calls you expect to make. Some services offer flexible pricing, so you only pay for what you use.

Pricing includes a setup fee, a monthly rental charge for the broadband line, a price per channel with a call allowance, and actual call charges. Choosing between metered or unmetered depends on the type of calls you make, like domestic or international. 

If you have multiple locations and use the same carrier, calls between sites are usually free. You only have to pay for the data circuits like broadband on a monthly basis. You can adjust your internet connection to match the maximum number of calls you expect to make.

The amount of bandwidth you need depends on whether you’re using voice, video, or messaging, and the codec being used. Fiber broadband offers high bandwidth rates, so it can support many concurrent calls. 

For larger businesses, SIP trunks are sometimes run over dedicated data circuits. The circuit supplier may offer a service level agreement or a quality of service guarantee. These ensure a certain level of service availability and prioritize voice traffic by providing a guaranteed amount of bandwidth and priority routing, or both.

This also gives you control over your business phone system. Meaning you can easily add new domestic or international locations without relying on a service provider to set up new infrastructure or connections. You can manage and control your own resources. 

SIP trunks are more reliable than traditional analogue circuits. They have number failover, so if the primary number doesn’t work, it can switch to a secondary number, even at a different location. Unlike traditional phone lines, they don’t have a fixed route for calls.

Voice over IP (VoIP) calls can take different routes to reach their destination, overcoming network issues or outages that would have stopped analogue calls. You can even install multiple broadband lines for extra resilience.

SIP trunking systems are cloud-based, which means you can access real-time analytics and data insights. This allows you to quickly see call logs and accurately measure usage and availability.

You can also use machine learning to analyze the data and gain valuable business intelligence. Plus, you can break down the statistics to see how individual employees are communicating with customers or prospects.

This real-time information can help improve employee performance and drive improvements in marketing, customer service, and sales.

SIP trunking allows teams to make calls, have video conferences, and send messages from one central location. In the past, these services were scattered, which made communication inefficient and costly. Once set up, managing your system is easy. By bringing all of your company’s digital communications together, you can streamline your communication processes.

SIP trunks make sure your phone calls have enough bandwidth for good quality. If your PBX isn’t overloaded, you’ll have clear calls even when it’s busy. AI technology can now improve call quality and fix errors like echoes and static. You don’t need to worry about dropped calls or echoey connections.

SIP trunks are different from older phone systems. They are dedicated connections for your calls. Your provider can keep an eye on your calls to catch any unusual activity. If anything unexpected happens, they will flag it right away. It’s a good idea to talk to your provider about the security measures they have in place. 

Lines can also be encrypted for an additional layer of security. This is because of the similarities of SIP to publicly available internet services, which can make it vulnerable to attacks. These can be prevented by going with a provider that secures both the data and audio layers with encryption.

SIP providers assign numbers to users. In essence, this allows you to have local numbers in any region, no matter where you are located. And if you decide to move, you can take your numbers with you to your new premises.

In the past, companies used ISDN circuits or copper lines that they physically installed on their premises. But now, there’s something called SIP trunking that replaces these analog phone lines with a digital version.

How does it work? Well, it breaks up calls into digital packets and sends them across a data network.

Let’s dive a little deeper and learn about the different components of the SIP trunking system. 

SIP trunks are made up of multiple lines, usually over 20. These trunks act as the middleman between a company’s phone connections and its Internet Telephony Service Provider (ITSP). Think of them as virtual phone lines that link a home or business to a telephony service provider.

They enable communication between the corporate network and various other phone and data networks. For instance, you can connect your PBX to the PSTN.

DID, or Direct Inward Dialing, is a telephone service that connects a block of phone numbers to your company’s PBX.

With DID, you can set up virtual numbers that bypass the main reception lines and go directly to a desk extension or group of extensions.

It works with local, premium-rate, or toll-free numbers.

Other names for DID include direct-dial numbers, direct dial, and direct dial-ins. So, when someone says, “reach me at my direct number,” they’re talking about DID. 

VoIP technologies need codecs to translate analog frequencies into digital data or one digital format into another.

Codecs convert audio voice signals into compressed digital formats for transmission over the internet. When the signal reaches its destination on the network, the codec converts it back to an uncompressed audio signal for replay. Basically, codecs are what make VoIP possible.  

There are different codecs with varying levels of compression. The two most popular codecs for SIP trunking are G.711 and G.729. G.711 gives you high-quality voice with no compression, while G.729 compresses the voice to use less bandwidth but sacrifices some sound quality.

Since SIP calls use your internet lines, call quality can be impacted by bandwidth. When you’re using data, you might not notice small delays in the signal or a bit of data loss. But for voice calls, even small delays can cause communication errors.  

Using the Quality of Service (QOS) setting on your router prioritizes calls and helps prevent errors by keeping audio quality high, even when other data-intensive activities are happening.

For example, if you’re downloading large files or streaming videos, a voice or video call can be given priority access to bandwidth while it’s happening. Once the call ends, the priority is lifted and downloads can speed up again. 

If you’ve decided that SIP trunking is right for your business, these are the three important things you to consider when looking at how to set it up:

When considering your service provider, you should consider whether your provider has their own network. This is important because private networks offer better quality control, faster recovery time, and overall better service.

Next, you should find out whether your provider proactively monitors their network, 24/7. This is crucial to ensure network diversity and uptime.

And lastly, if a network failure happens, can your provider seamlessly reroute calls? It’s important to know what measures they have in place for business continuity.

Configuring the PBX can vary based on the software and provider you use. To get started, you’ll need some credentials from your provider, like a username and password, server hostname or IP address, and SIP port number. Once you have these, you can enter them into the PBX configuration settings.

When implementing SIP trunking, it’s important to consider security. Businesses need to configure their firewall and network security settings to allow voice traffic.

They should also use encryption and authentication to protect against unauthorized access and eavesdropping.

Firewall configuration can be complex, so it may be helpful to consult with IT or a network security professional. SIP encryption and authentication can be done using TLS and SRTP, which encrypt and authenticate traffic. These settings can be configured in the PBX and IP phones.

These are some best practices and things to keep in mind when considering setting up SIP trunks for your organization:

Audio is crucial to call quality and any disruptions can negatively impact communication. With all your data traffic competing for limited bandwidth, you need a way to prioritize specific traffic types.

Thankfully, QoS settings on your router let you prioritize SIP traffic for audio call quality over less demanding data traffic types.  

If your network goes down, it could disrupt your business operations. However, with SIP trunking, you can avoid this problem since it allows calls to be seamlessly rerouted in case of a network failure.

This means that even if one line fails, your calls can still be routed through another line. It’s like having multiple hard-wired lines for a traditional phone system.

Make sure your SIP provider takes advantage of this feature to ensure uninterrupted communication for your business. 

The flexibility SIP trunking affords businesses helps them to quickly scale their businesses up or down as needed. As your operation grows, you can add more numbers to accommodate your expanding workforce.

In addition, SIP trunking allows you to retain numbers irrespective of your geographical location. This means you can quickly scale globally, expanding your operations into new countries – while keeping your existing numbers. 

Organizations have been using SIP calls for years. This is because it helps them save money while offering additional benefits like reliability, scalability, and customization.

Let’s look at some business use cases across industries:

Retailers use SIP trunking for efficient customer service, including managing call volumes during peak shopping seasons.

Contact centers leverage SIP to handle high call volumes, improve call routing, and enhance customer interactions.

Healthcare organizations use SIP for telemedicine, appointment reminders, and seamless communication among staff.  

Banks and financial institutions rely on SIP trunking for secure client communication, and reducing operational costs.  

Large enterprises benefit from SIP trunking for unified communications, remote work support, and international calling. 

While SIP trunking has many benefits for businesses, there are also a few common challenges that businesses can face. These are some challenges you can expect to face – and how to deal with them: 

Call quality is a common issue. Since the technology relies on your internet connection to transmit your voice and video data, factors like network congestion, latency, jitter, and packet loss can impact the quality and clarity of your calls.

To ensure good performance, you need enough bandwidth for peak traffic and to prioritize voice and video traffic over other data. It’s also important to monitor your network and troubleshoot any issues that come up. 

You can calculate how much bandwidth is needed by identifying the maximal number of concurrent calls for your organization at any point. Then you can multiply this by the bit rate your codec uses.

Security represents another challenge businesses might encounter. SIP trunking exposes your phone system to the internet, making it vulnerable to cyber attacks like DoS, spoofing, eavesdropping, fraud, and theft.  

To prevent unauthorized access and data breaches, you must implement strong security measures like encryption, authentication, firewall rules, and anti-virus software. It’s also important to comply with relevant regulations and standards in your industry and location, such as HIPAA, PCI-DSS, GDPR, and others. 

When setting up SIP trunking, one of the first challenges is making sure your phone system and provider are compatible.  

Not all phone systems support SIP trunking and not all SIP providers offer the same features and protocols. You might need to upgrade your phone system, add new hardware or software, or adjust your network settings.

Make sure to also check the compatibility of your codecs, firewalls, routers, and other network devices that can impact the quality and security of your voice and video traffic. 

Configuring SIP to work with other applications and systems can also cause hiccups. SIP trunking allows you to use voice and video services like unified communications, collaboration, conferencing, messaging, and analytics.

But it’s important to make sure your system can work with these applications and systems, exchanging data and functionality seamlessly.

You might need to use APIs, gateways, or adapters to connect your system with other platforms and devices. 

This is a useful guide to help you navigate commonly used terms related to SIP trunking: 

A private branch exchange (PBX) is a phone network used within a company or organization. Its main purpose is to enable employees to connect with each other internally and make/receive external phone calls. 

PSTN is short for Public Switched Telephone Network. It’s a global network of phone systems that use traditional circuit-switched networks for public communication.

You may also hear it referred to as landlines, POTS, or fixed-line telephones. This system has been around since the late 1800’s and is still widely used today.

VoIP, or Voice over Internet Protocol, is a way to make calls through the Internet. Some people also call it Voice over IP.   

To set it up, you need a router for the internet and a handset connected to it. Instead of using a physical landline, the handset uses the internet to transmit your speech.  

Phone providers use VoIP to provide telephone services to customers without the need for physical phone lines and equipment. VoIP allows for high-quality calls and offers advanced features that you don’t usually get with a landline.  

A Session Border Controller, or SBC, is a device that protects and regulates IP communications. SBCs are placed at network borders to control IP communication sessions. They were initially designed for VoIP networks but are now used for all types of real-time communication, such as VoIP, IP video, text chat, and collaboration sessions.

SIP trunking offers a flexible, cost-effective, and scalable alternative to traditional phone lines. Armed with this SIP Trunking 101 guide, you’re well on your way to unlocking the full potential of this technology for your organization.