
Determine How Much Internet Bandwidth You Need for SIP Trunking in 3 Steps
Although SIP trunking is not complex and can be deployed by organizations with very little networking or telephony experience, there are a few key factors that will determine performance and reliability. Chief among these is the amount of Internet bandwidth available to carry voice traffic to and from the SIP provider. By using the session initiation protocol (SIP), SIP trunks establish, manage, and terminate calls seamlessly, providing scalable and flexible voice communication services over the internet. So, it is wise to spend a bit of effort figuring out exactly how much bandwidth you will need to support your business. Fortunately, it is fairly easy to do.
What is SIP Trunking?
SIP trunking is a method of delivering voice over internet protocol (VoIP) communications between a business’s private branch exchange (PBX) and the public switched telephone network (PSTN). This technology allows multiple phone calls to be made over a single internet connection, effectively eliminating the need for traditional phone lines. By using the session initiation protocol (SIP), SIP trunking establishes, manages, and terminates calls seamlessly.
With SIP trunking, businesses can make voice calls over the internet, which significantly reduces the reliance on traditional phone lines. This not only streamlines communication but also offers a more cost-effective solution for managing voice calls. By leveraging internet protocol (IP) for phone calls, companies can enjoy greater flexibility and scalability in their communication infrastructure. In essence, SIP trunking modernizes the telephone network, making it more efficient and economical.
Understanding SIP Trunk Channels
A SIP trunk channel is essentially a line for a single call on a SIP trunk. Each SIP channel is required for each concurrent call, utilizing the internet connection to link to the necessary networks for connecting calls to receiving phones. While the terms SIP channels and SIP trunk lines are often used interchangeably, some prefer “SIP trunk line” for consistency with traditional telephone terminology.
The number of SIP trunk channels you use directly impacts your telephone bill. Most SIP trunk providers sell channels in batches, and the cost per channel typically decreases as you purchase more. This scalability allows businesses to adjust their communication capacity based on their needs, ensuring they only pay for what they use. Understanding how many SIP channels your business requires is crucial for optimizing both performance and cost.
Step 1 – Determine the Number of Concurrent Calls You Must Support
Since each inbound or outbound call requires a certain amount of bandwidth, the first step in calculating your needs is to figure out how many calls are likely to occur at any one point in time. For most companies, this is not equivalent to the number of employees or extensions because not everyone is on the phone all the time. Most businesses run about one concurrent call for every three or four employees. So, if you have a 10 person company, in most cases you can expect 3 – 4 simultaneous calls. Of course, this changes if your company has a very call intensive business model, such as a call center or an active inside sales team, so you’ll need to factor in both the number of employees and the type of telephone work they do.
In addition, to being an important calculation for determining your bandwidth requirements, each SIP channel supports one concurrent inbound or outbound call, so this assessment will be necessary for determining the number of SIP trunking channels you will need. Properly sizing your SIP trunk connections is crucial to avoid issues such as blocked calls and poor service quality.
Step 2 – Ask Your SIP Trunk Provider about the Codec
The Internet bandwidth necessary to support each SIP trunking call depends on something called the voice codec used by your SIP trunking provider. You don’t have to know too much about the technicalities of the voice codec (check this out if you’re curious), but you do need to ask your SIP provider which codec they use and how much bandwidth it consumes per call. SIP.US uses the G.711 voice codec, which consumes 85kbps of bandwidth per call.
It’s important to note that SIP lines and SIP trunk channels are essentially interchangeable terms, both referring to individual call connections within a SIP trunk.
Step 3 – Do a Little Math
Simply multiply the expected number of calls by the per call bandwidth requirement given to you by your SIP vendor (in our case 85kbps per call) and you’ll know the minimum amount of bandwidth you require. Keep in mind that other data traffic will likely be traversing your connection during calls, so leave a margin of at least 10%. Now all that’s left to do is determine the amount of bandwidth available.
In order to measure Internet bandwidth, a speed test should be run from a computer that utilizes the Internet connection to be tested. There are many speed tests available online, one example is http://speedtest.net.
You’ll want to use the lower of your upload or download speed (almost always upload) for the final calculation. Speed is usually presented in Megabits per second (mbps). To get Kilobits per second (kpbs), multiply mbps by 1000. Now just subtract the number of kbps required from your bandwidth and you’ll know if it is sufficient. Here’s an example:
Number of expected concurrent calls for a 10 person business: 3
SIP vendor’sper call bandwidth requirement: 85kbps
3 x 85 = 255kpbs
Upload speed = 2.55mpbs
2.55 x 1000 = 2550kbps
2550/85 = 30
This connection (which happens to be high speed cable) is more than sufficient to support 3 concurrent calls. In fact, it could support 30.
There you have it, the secret to determining the number of calls your Internet connection can support. We find that most companies with cable, DSL, T1 or metro Ethernet connections have plenty of bandwidth to support their needs. Check out this post for a details on a few more considerations that will impact your SIP trunking experience. As always, feel free to contact us if you have any questions or would like help evaluating your specific needs.
Choosing the Right SIP Provider
When selecting a SIP provider, several key factors should guide your decision:
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Quality of Service: Ensure the provider offers high-quality voice calls with minimal latency and packet loss. This is crucial for maintaining clear and reliable communication.
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Reliability: Choose a provider with a robust network and a proven track record of uptime. Consistent service is essential for business operations.
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Scalability: Opt for a provider that can grow with your business, offering flexible pricing plans and easy upgrades to accommodate increased call volumes.
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Cost: Compare prices among providers, considering the cost per channel and any additional fees. Look for a balance between affordability and quality.
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Customer Support: Excellent customer support is vital. Look for providers that offer 24/7 technical assistance and a user-friendly control panel.
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Features and Functionality: Consider the features you need, such as toll-free support, domestic and international calling, and E911 support.
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Integration with Existing Systems: Ensure the provider’s SIP trunking services integrate seamlessly with your existing PBX and other systems. This will facilitate a smooth transition and ongoing operations.
By carefully evaluating these factors, you can choose a SIP trunking provider that meets your business needs and ensures a reliable communication infrastructure.
SIP Trunking vs Traditional PRI-ISDN
SIP trunking offers several advantages over traditional PRI-ISDN, making it a superior choice for modern businesses:
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Cost Savings: SIP trunking eliminates the need for traditional phone lines, reducing costs and providing a more cost-effective solution for voice organization and termination.
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Flexibility: Unlike PRI-ISDN, SIP trunking allows businesses to scale up or down as needed without requiring physical infrastructure changes. This flexibility is ideal for growing businesses.
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Improved Voice Quality: SIP trunking provides high-quality voice calls with minimal latency and packet loss, ensuring clear and reliable communication.
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Increased Reliability: SIP trunking offers a more reliable solution than traditional PRI-ISDN, with fewer outages and disruptions. This reliability is crucial for maintaining business continuity.
By transitioning to SIP trunking, businesses can enjoy these benefits and enhance their overall communication strategy.
Troubleshooting Common SIP Trunking Issues
While SIP trunking offers many advantages, businesses may encounter some common issues. Here’s how to troubleshoot them:
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Poor Voice Quality: Check for packet loss, latency, and jitter. Ensure your internet connection meets the required bandwidth standards and consider implementing quality of service (QoS) settings to prioritize voice traffic.
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Dropped Calls: Verify that your SIP trunk provider is not experiencing outages or technical issues. Check your PBX configuration to ensure it is set up correctly and that there are no network issues.
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Inbound Call Issues: Ensure your SIP trunk provider’s configuration is correct for routing inbound calls. Verify that your PBX is set up to receive inbound calls and that there are no firewall or network issues blocking the calls.
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Outbound Call Issues: Confirm that your SIP trunk provider is not blocking outbound calls. Check your PBX configuration to ensure outbound calls are being routed correctly and that there are no network issues.
By understanding these common issues and taking proactive steps to troubleshoot them, businesses can ensure a smooth and reliable SIP trunking experience.