What is SIP Trunking?
SIP (Session Initiation Protocol) Trunking is a digital way of making and receiving phone calls and other communications over an internet connection. The term trunking refers to the method of consolidating multiple communication channels into one singular connection. SIP Trunks provide VoIP (Voice over Internet Protocol) connectivity between on-premise phone systems to the PSTN, which allows for the efficient use of resources and connection to the telephone network.
Working with a trusted SIP provider like Fuse 2 ensures your VoIP infrastructure is built with security at its core, not as an afterthought.
Understanding SIP Call Quality Problems
Understanding SIP call quality is relatively simple when you remove the jargon! Many providers mention words such as jitter, latency and packet loss (see Cisco’s explanation of jitter, latency and packet loss), which can sound confusing; however, they’re quite simple to understand.. SIP runs over the internet; if your connection is unstable, you will experience inconsistencies in your calls.
Jitter and latency relate to the actual quality of the call; this means if your connection is unstable, you will experience scrambling of words, pauses in audio and overall poor voice quality. Packet loss, on the other hand, means that your audio can completely cut out mid-conversation if the connection is lost or unstable.
Jitter and latency are the most common challenges you will face if your provider has insufficient bandwidth. Packet loss, however, is classed as a P,1 and if this is happening regularly, it’s time to look for a new provider!
The Most Common Causes of Bad SIP Audio
Insufficient Internet Bandwidth
VoIP and SIP calls require steady upload and download speeds for high-quality calls. If your network’s bandwidth is full or there is a network instability, your call quality falls back onto the lower-bandwidth format to not drop the call. It is crucial for your provider to test your bandwidth regularly and to notify you should they believe that you are reaching your bandwidth capacity.
Network Congestion
If your network is congested with other devices which are using too much bandwidth, this leaves little room for voice calls. It is always advised to minimise bandwidth-intensive activities such as streaming to free up these channels for voice data.
Poor Router or Firewall Configuration
If your firewall or router has been set up poorly, you will experience an array of issues. One of the most common in this case is one-way audio, meaning you can hear the caller, but they can’t hear you, often caused by SIP ALG issues. This is caused by incorrect port forwarding or NAT traversal issues. Working with a reputable SIP provider to review your configurations will resolve these issues going forward.
Wrong Codec Settings
Inefficient, poorly coded or incorrect codec settings can significantly impact voice quality. If your codec settings have been wrongly configured, you will experience quality issues such as voices sounding muffled, distorted or robotic. Your provider should know what codec settings are appropriate for your usage and can come across trade-offs in the process, such as G.711 offers higher call quality but uses more bandwidth; G.729 uses less bandwidth but can sacrifice quality. This decision will be made by your provider beforehand based on pre-requists when onboarding.
Provider Issues
Your SIP stability is only as good as your provider’s network and the connectivity in your area. Issues may appear higher up the chain in the provider’s SIP network and in this case, it is your provider’s responsibility to address this. It is important however, to choose a provider who offers strong SLAs and who proactively monitor their network architecture 24/7 and have full visibility of all call quality. In the purchasing phase of your customer journey it is vital to test other SIP providers as their network may work more cohesively with your set-up.
How to Troubleshoot Bad SIP Call Quality
When it comes to troubleshooting bad SIP quality, the first thing you need to check is internet connectivity and bandwidth. If your assessments reveal unstable connectivity, purchasing a business-dedicated broadband with suitable bandwidth will be a crucial investment.
Using speed test tools to analyse connection metrics during calls will also enable you to be at the forefront of issues; high jitter and latency can indicate that there are issues in your setup. It is also important to carry out regular internal testing upon go-live – the last thing you want to do is go straight out ringing your customers when your telephony set-up may not work! Internally, these are troubleshooting tasks that you, the customer, can conduct yourself. However, in the background, your provider should be testing high-level configurations such as QoS settings, firewall configurations and bandwidth capacity.
Proven Fixes for Better SIP Call Quality
While there are some technical problems that require experts in the field to fix, there are some best practices that you can follow to ensure better call quality. These are:
- Upgrading to business-grade internet: sometimes standard internet doesn’t quite cut it for VoIP. Investing in dedicated business broadband can greatly enhance your connectivity’s reliability.
- Choosing Wired over Wireless Connectivity: It is proven that Ethernet (wired) is more reliable and faster than Wi-Fi. Ethernet offers higher bandwidth and lower latency – perfect for businesses that want to adopt VoIP into their business.
- QoS (Quality of Service) Testing: Ensuring QoS tests are conducted regularly is vital in ensuring your traffic is being prioritised correctly. This ensures that critical applications such as voice receive the necessary bandwidth even when the network may be congested.
When to Involve Your SIP Provider
It is common in VoIP that issues are an upstream issue directly from the provider/carrier. In this case, it is your provider’s responsibility to proactively mitigate these issues, reaching their customers and affecting their communications. It is crucial to choose a provider who has full visibility of their network and can easily identify abnormalities in call quality before they escalate. To identify upstream carrier issues, you can visit websites such as downdetector.co.uk. It is a legal requirement that carriers are fully transparent about network outages,s and here you can find out if it is an upstream issue in seconds. Unfortunately, if an issue has occurred upstream, you are unable to assist in resolving this. Your provider, however, should be proactively updating you throughout the outage and should be assessing all options to get your services live again. If your provider is sitting back waiting for the storm to pass, it’s time to make the change!
How Fuse2 Can help your business
When it comes to VoIP and SIP, it is important to stick to prominent providers (like Fuse 2!) who display strong SLAs, advanced technical support and heightened security. Rather than waiting for issues to inevitably occur and cost you more money, now is the perfect time to assess your options!
For a free SIP quality check and a migration incentive plan – get in touch today for a full, tailored proposal!
