Best Ways to Troubleshoot & Improve VoIP Call Quality
Trying to improve sound quality during VoIP calls? Learn how VoIP call quality is measured, troubleshooted, and easily improved.
The pressure to exceed customer expectations is on for enterprises. Nearly 58% of consumers will end ties with a brand or business due to poor customer service. This means your VoIP call quality must be top-notch to enable your agents to provide the best customer service possible without having to repeat themselves or risk a dropped call. For customers to perceive your business as stable, reliable and capable, your VoIP voice quality must exhibit the same characteristics.
In this guide, we’ll explore several facets of VoIP call quality and ways to improve it. We’ll introduce common voice quality problems, how to test and measure call quality using Mean Opinion Score (MOS) tools and codecs for prioritizing your telecommunications bandwidth.
Finally, we’ll review call quality best practices and the benefits of using call analytics to achieve an impactful and productive conversation.
Understanding VoIP Call Quality
Call quality is the measure of the efficiency and effectiveness of conversations between a customer service representative and customers. When it comes to VoIP, call quality refers not only to the attitude and competency of the agent on the line but the audio and video quality of the call as well. Customers expect a smooth, reliable connection during which they can clearly hear every sound and, if video is included, see every move.
Calling is a way customers can get information and relay their brand experience quickly. So it’s important to ensure a satisfactory VoIP call experience to maintain the level of trust and customer experience they expect of your company.
Quality of Service (QoS) and Its Impact on Call Quality
Quality of service (QoS) is technology that sorts and prioritizes your telecommunications traffic to reduce and prevent data degradation. QoS standards tell your telecommunications network which data should be sent first and how much bandwidth to dedicate to it so that it arrives on time and in one piece.
When you and your team are holding video calls, sending large files, streaming audio or video, or otherwise maintaining a high level of non-call-related activity on your network, you’re taking up bandwidth. That negatively impacts the clarity of VoIP calls because audio drops down to the lowest, but most reliable format your network’s available bandwidth will allow as to prevent call drops. QoS works with your SIP trunk and direct routing like a traffic manager, ensuring the data sent by your company is in top-quality condition and delivers seamlessly.
Without QoS in place, data is sent and received in a first in, first out (FIFO) order which can cause major backups and delays in sending/receiving larger files or streams.
Most Common Voice Quality Issues
To really understand voice quality and improve the volume of reliable calls your team is able to make, you need to know about the most common VoIP quality issues. There are three VoIP network performance issues all call centers and large businesses run into: latency, network jitter, and packet loss. Let’s briefly explore all three:
Have you ever been on a video call where the other person’s lip movement doesn’t quite match up to what you’re hearing them say? This is VoIP latency. It’s a delay in the transmission of audio and/or video from your call. It’s usually caused by network congestion, though it can also result from poor wiring or outdated network equipment. Upgrading your team’s hardware and software while reminding them to reduce other bandwidth usages as possible can reduce latency. Having your IT team implement QoS standards can also help.
Benchmarking latency levels
As companies developed digital networks in the 1980s and 1990s, and VoIP became possible, many businesses rushed to adopt the technology to take advantage of its global reach and relatively lower cost. The International Telecommunication Union (ITU), which standardizes telecommunications and information communication technology for all countries, established the G.114 recommendation for acceptable voice delays. G.114 states that 150 milliseconds is an acceptable delay for transmission of speech and other audio. The longest delay it will approve in new tech is 250 milliseconds.
2. Network Jitter
Similar to latency, network jitter occurs when packets arrive at their endpoint out of order because of network delays. VoIP transforms audio information into digital units called packets to transmit that information to its destination. There are three main parts of a data packet:
- The header instructs the endpoint about the length, type, and origin of the data in the packet.
- The payload or body of the packet. This is the data itself.
- And the footer checks for errors in transmission and tells the endpoint that this is the end of the packet.
Latency is a simple delay in packet delivery. Jitter is a bunch of delays of varying lengths in packet delivery. Normally, packets travel at standard intervals with 10 milliseconds in between. But data starts to feel “jittery” when the network becomes congested, or the IT staff monitoring it hasn’t prioritized traffic on it. It looks like a flickering video that goes in and out. It sounds like a conversation when you hear the other speaker fine for a few words, but then they cut out for a few seconds. Jitter causes big headaches for companies that stream online games or video because users won’t tolerate a disjointed experience in those mediums for long.
What’s an acceptable level of network jitter?
Some jitter is inevitable as networks continually prioritize different types of data transmission as well as their user count. That said, you should aim for no more than a 150 millisecond one-way delay in packet delivery. You can conduct a latency test/jitter test by using the jitter test tools in your network’s admin or reports section. Jitter tests send thousands of ping measurements to servers across the globe to measure their speed. Jitter test results can offer insights into how to prioritize data and improve your overall quality of service.
3. Packet Loss
Packet loss happens when one or more packets of data traveling between two endpoints of a digital call fail to reach their destination. Sometimes packets arrive late and the file or stream isn’t noticeably worse, but packet loss still has a significant economic effect: Statista reports that in 2017, packet loss caused surveyed companies between $301,000 and $400,000 in lost revenue. Packet loss is caused by overtaxed networks, outdated software or session border controllers, or denial-of-service (DoS) attacks.
Just like improving latency and network jitter, to reduce packet loss you need to prioritize VoIP traffic and phones. Ensure everyone in your call center is using quality network equipment, sticking to professional activities only, and compressing any files they send. Furthermore, you’ll want to encourage your team to use dedicated ethernet lines for VoIP calls as much as possible. WiFi isn’t as reliable for call quality, especially when dozens or hundreds of calls are happening at once over the same WiFi network.
Acceptable packet loss rate
What’s an acceptable packet loss rate? It depends on what you’re transmitting to your audiences. Packet loss around 1% is satisfactory for streaming audio or video; you won’t notice much difference in VoIP call quality. A 1-2.5% packet loss rate is acceptable. Packet loss rate from 2.5-5% will result in noticeable delays and drops. Telecommunications experts consider any loss rate above 5% unacceptable.
For all three call quality issues, you can work with your cloud communications provider to proactively test call quality issues. You can monitor your bandwidth usage, carriers, routes and error codes to identify pain points and avoid costly downtime.
How is Call Quality Measured?
Mean Opinion Score (MOS) is a rubric used to grade the audio quality of phone connections. To calculate call quality via MOS, a group of listeners judge the quality of an audio sample on a sale of one (the worst) to five (the best). This group often self-selects from people who agree to complete a quick survey about call quality at the end of their call with a business. The average score of all listeners in the test is the MOS for that system, and the codec.
You may be wondering, what’s a good call quality score? That also depends, this time on how your company has performed in the past, your industry standards, and your ability to improve on a given timeline. Because most mean opinion scores are rated on a scale of 1 to 5, 3 is an average score. But experienced companies know that they need to aim for a MOS of 3.5 or higher, and ideally, a MOS of 4.3 or more. Below a score of 3.5, customers begin to rate call experiences as unacceptable.
We should note that there’s much more to measuring call quality than assessing the audio. Check out our guide to call center metrics and get the full rundown on top call quality KPIs like customer satisfaction, first call resolution, and net promoter score.
VoIP Codecs for Better QoS & Call Quality
If you don’t have a background in the mechanics of data transmission, how can you ensure you’re setting up your QoS to prioritize and send data properly? Get to know codecs.
Codecs are rules about the amount of data compression that happens when transmitting your VoIP audio and video, if applicable. These rules are based on the available bandwidth, your desired audio/video quality, and sometimes, your network equipment. Codecs are named because they compress your analog audio and video into transmittable digital data through your available bandwidth, then decompress it to transform back into viewable video and audible sounds. Another way to think about them is that they code sounds (and other transmissible data) at a certain rate that balances quality with speed.
It’s important to determine properly calculate the expected number of calls by the per call bandwidth requirement given to you by your SIP vendor. This will let you know the minimum amount of bandwidth you’ll need. There are three main codecs that VoIP networks use:
- G.711. Many telecommunications professionals consider the G.711 codec to be optimal for VoIP voice quality because it doesn’t digitally compress data. This codec’s high bit rate of 64 kbps offers great call quality. It’s often the default codec for SIP trunk providers and can be used anywhere that enough bandwidth is available. As a bonus, there are no licensing fees to operate with the G.711 codec.
- G.722. This codec’s transmission rate can transmit audio at 48, 56, or 64 kbps. It’s free for businesses to use, and it’s a preferred codec for VoIP users as well as radio broadcasters. G.722 is a wideband codec, meaning it covers audio frequencies from 50 Hz to 7000 Hz at a superior quality level over the G.711 codec. It’s a go-to way to transmit everything from recorded meetings or demos to music to simple conversation. G.722 rarely has issues with latency or network jitter.
- G.729. The G.729 codec operates on a narrow band of the audio frequency spectrum and transmits at 8 kbps. While it’s great for larger volumes of simple audio VoIP calls, it’s not meant for high-quality audio transmission like music. Your business would need to make sure that other online activities don’t encroach too much on your VoIP and further reduce the quality of calls sent via this codec.
Knowing which codec is right for your communications activity will go a long way toward ensuring a high quality voice call experience for your audiences.
4 Call Quality Best Practices
Achieving better call quality usually means adjusting the quality of service settings for your system and discovering the proper codec for your data needs. In most cases, adequate bandwidth and the proper codec is enough to ensure high call quality. But if you are still having problems with delay, jitter, or packet loss, you will want to examine your overall call quality practices.
90% of customers say that customer service is an important factor in the brands they do business with. If those brands fall short of expectations, they have no problem cutting ties. Don’t let your customers walk away and decrease your lifetime value (LTV) benchmark because of poor voice call quality affecting their experience. Here are four call quality best practices:
1. Test your internet connections regularly
Knowing how often your network jitters, delays, and loses packets is key to improving voice quality. The strength of your Local Area Network (LAN) and Wide Area Networks (WAN) have a huge impact on how seamlessly you can send and receive VoIP calls.
2. Prioritize voice traffic on your VoIP system
Nothing is more frustrating than being on a call with someone and not being able to understand them because of a bad connection due to overcrowded bandwidth. That file or image you need to send a customer should be deprioritized in favor of live demos, training calls, and customer service interactions. Have your IT administrator use QoS to prioritize voice traffic on your network. This sends a subtle message to customers that your live connection matters most.
3. Use SIP monitoring tools
Your Session Initiation Protocol (SIP) governs the origin and termination points of your VoIP calls. It directs phone traffic along your SIP trunks and yes, it can be another cause of VoIP bandwidth problems. Luckily, you can use SIP monitoring tools to analyze why VoIP call failures happen, how to better prioritize call traffic, and generate alerts so you can fix issues right away.
4. Work with a reputable VoIP provider like AVOXI
A good VoIP provider will not only help you set up a unified communications network but actively advise you on how to troubleshoot your system and resolve bandwidth issues. They’ll have highly refined QoS standards and will help you prioritize the specific kinds of data you need to achieve your business goals.
Automatically Troubleshoot Voice Quality Issues with Call Insights
The most effective way to improve your voice quality is to consistently monitor your call quality with analytics. A self-service dashboard of VoIP monitoring tools can help your team proactively monitor voice quality to ensure your call center stays well-connected to your customers with as little downtime as possible. Although a call insight dashboard takes some time upfront to set up and learn, over time you’ll be able to:
- Analyze any time. Support teams don’t always have the same schedule as you, especially if they’re in another time zone. A self-service call insights dashboard lets you and your remote teams examine your call quality analytics and flag potential areas for improvement at your convenience.
- Enhanced call quality visibility. Having greater visibility into your call center’s call quality stats helps ensure business continuity across teams and regions.
- Operate more efficiently. Don’t waste time having your agents type out lengthy explanations of a voice quality problem. Create support tickets using data you pull directly from your call insights dashboard, and link tickets about related call quality issues. Then get back to those tasks that drive your bottom line!
- Improve your customer experience. Greater insight into call quality analytics allows your business to take targeted action that reduces downtime, lags, and jitters. Such proactive activity results in a better customer experience.
- Real-time monitoring. When you have the ability to monitor and instantly report call problems, you can start improving call quality right away.
Explore High-Quality VoIP Routes in Over 170 Countries
VoIP has revolutionized the way businesses connect with customers around the world. But what hasn’t changed is the first impression you make when speaking with a customer. Set up your business for successful VoIP interactions on every call by actively monitoring your call insights. When you know how your network is performing, you can apply those call insights to make your customer interactions even more impressive.
Want more information? AVOXI offers a 60-day free trial of call insights whenever you buy or port a phone number. Our knowledge powers our software-driven, cloud-based communications that provide outstanding coverage across more than 170 countries. We offer the best VoIP call quality so you can hold crystal clear conversations with reliable network continuity. Visit AVOXI.com to learn how we can help you deliver superior customer experiences.
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