How SIP Trunks Help to Ensure Consistent Call Quality

Posted on: 2017-01-24 | Categories:SIP

When mobile phones first became popular, call quality was a big issue. Most people felt that landlines had better audio than using a cell phone. Over the years this difference in quality has steadily decreased to the point where cell phone quality is acceptable for calls. Even business users are generally satisfied with the audio quality of mobile devices.

When VoIP became widespread, the same cycle continued. At first people associated VoIP calls with poor audio. But advances in VoIP technology and proper implementation have ensured that VoIP calls are also crystal clear. With mobile devices and landlines, call quality largely depended on the carrier.

When it comes to SIP trunks however, you can control call quality. It is easy to overlook clarity and reliability when you are more focused on reducing costs or getting the latest features. But all that becomes meaningless if users are unable to hear each other or calls are dropped frequently. So what can you do to make sure that your SIP trunks give you consistent call quality? There are a few things you can do to make that happen.

Increase the Available Bandwidth

SIP trunks route voice calls over your company’s data network and the Internet. There is no separate network to maintain or wiring to install for your phones. It is one of the advantages of switching to SIP trunking. Although this does mean voice calls join all the other types of media jostling for space on the same Internet connection.

So if there isn’t enough bandwidth for data, your voice calls may degrade substantially. Users might miss words or hear echoes. Calls may not go through or they get dropped in the middle of conversations. You certainly don’t want that to happen when an employee is talking to a customer!

One way of resolving the issue – especially if your network is frequently overloaded – is to increase the speed and bandwidth of your network. If there is more bandwidth, voice calls don’t have to go to the back of the queue. SIP trunks now support HD voice, which offers even more clarity than ever before. As long as you have sufficient bandwidth, SIP trunks will give you exceptional audio.

Configure QoS Correctly

Suppose your network is not the problem but your calls are experiencing issues. What’s the problem then? This is where Quality of Service comes into play. You can configure QoS on a hardware device which is usually a router or firewall. Basically it tells the network that voice calls take priority when they are in a queue. If a call is waiting along with a document and some other media at a certain point in the network, the voice packets will go through first. Voice calling is more sensitive to time delays than software applications, so having proper QoS is essential.

Sufficient bandwidth capacity and QoS are the twin pillars of voice quality when it comes to SIP trunks. However, there is another factor that affects call quality, though you have little control over it.

Compression and Codecs

Most data that travels through the Internet is compressed before transmission. If you try to transmit all data without any compression, it would severely strain the network capacity. The vast majority of organizations do not have the infrastructure to support it and neither is it necessary. There isn’t much of a perceptible loss of quality when you compress most data and the same is true for voice calls.

In the world of SIP trunks, a codec is a piece of software that converts audio signals into digital data before transmission. At the other end, the codec converts the packets back into analog sound signals so you can hear the words normally. The codec may also compress the voice signals in addition to converting it into digital format. You should ask your vendor about codecs before purchasing your trunks since you won’t be able to change it later.

Some codecs are specially developed to use as little bandwidth as possible. So you can make several concurrent calls on limited bandwidth, although quality will suffer a bit. Others codecs do the opposite – minimize the amount of compression so voice quality is better. But you will need more bandwidth to accommodate several calls at once. Some codecs offer high bit rates for even better quality but they are useful only for voice critical applications in very few networks.

SIP trunks give you better quality than even landlines today, a far cry from the early days of poor audio.You should monitor your system in real time as well, so you can fix problems if something changes in the environment. Once you have optimized your hardware and network for VoIP calls, you will get consistently high quality audio calls.