nexVortex BLOG

Don’t Let Your Call Quality Suffer

Don’t settle for poor voice quality.

Patti Dean

Don’t settle for poor voice quality.

In the early days of Voice over IP (VoIP), poor call quality was rampant.  There are several very valid reasons why this was true. When VoIP was first rolled out, bandwidth was at a premium; meaning not only were big pipes into enterprise buildings relatively scare, they were also very expensive. This introduced congestion in the “last mile” where multiple services (voice, data, and video) all competed for available bandwidth. The last mile is the physical path between the network and the building over which services are delivered. If it is too small, it is a choke point.  Secondly, VoIP calls are carried over the public internet where there is no guarantee that packets will take the same path (between the sender and receiver), no guarantee that all the packets will actually make it to their destination (packet loss), and no guarantee that the time interval between packets arriving will be consistent (jitter).  These phenomena lead to latency, packet loss, and jitter which are killers of voice quality. The industry made great strides over the years.  As fiber buildouts continued into enterprise buildings and as competition drove down price, the pipes in the last mile got larger and less expensive.  The internet itself continued to evolve as carriers upgraded their networks with bigger and faster routers and bigger and better transport in the backbone. But that wasn’t enough Call quality can be measured on a Mean Opinion Score, or MOS. The score is a 1 to 5 scale with 1 being an unacceptable score to 5 being an excellent score.  A good quality call would have a MOS score in the range of 3.8. A problem with maintaining an acceptable level of quality is that there are many links in the chain between the calling party and the called party.  There is the Local Area Network (LAN) in the building (carrying a multitude of traffic); there is the access network (made up of any number of carriers/ISPs) carrying traffic to the core and then there is the core of the internet itself. As voice packets leave the building, move across the access network and hop their way across the internet to their ultimate destination, they can be handled by many, many carriers.  In fact, studies have shown that IP packets are handled by an average of 3.5 carriers as they move from point A to point B. This raises the question – with so many potential parties involved in the handling of a call and with a lack of control over the path every packet takes – how can voice quality be guaranteed?  How do you tell where and when quality is being negatively impacted?

Well, you couldn’t – until nexVortex deployed mSIP.

What is nexVortex mSIP and how does it work? mSIP stands for Managed SIP Trunking and it is as unique as it is powerful.  The nexVortex service is delivered with an on-premise router (enterprise session border controller) which provides a demarcation point on the premise as well as a point to measure voice quality before it ever leaves the building.  To ensure that packets can transit to their destination un-encumbered by the troubles of the core internet, our service carries those packets — not across the public internet — but rather across our nationwide private network which can guarantee quality. To ensure that packets get delivered directly to us from the ISP which the business customer chooses to use, we executed direct peering relationships with most of the major carriers. That means voice packets get handed directly to us by that ISP thereby reducing hops and hand-offs which in-turn reduces latency and jitter. That means better voice quality. However, that is only part of the solution.  We take voice quality measurements at multiple points along the call path and we stitch call quality data together using powerful data correlation techniques and then utilize data visualization and reporting tools to allow mSIP users to see where quality is being impacted, when it is being impact, and how badly it is being impacted – all in real-time. The solution is shown below.



Managed SIP Trunking (mSIP) sets the bar Even though the industry made strides over years by improving last-mile bandwidth size and price, as well as continuing to evolve the core of the internet, nexVortex knew that wasn’t enough.  We wanted to do better.  We made the investment, did the development and deployed mSIP to deliver Quality, Available, and Visibility for SIP Trunking. It is one of the fastest growing offerings in portfolio and is recognized by channel partners and business users for the tremendous value it provides. If you want to work with a company focused on delivering innovative solutions like mSIP which address critical business communication needs, contact us today to see how we can help your business.


Considering a Move to Microsoft Teams for Voice? – We Have Your Migration Strategy

Microsoft Teams users can immediately take advantage of nexVortex’s feature-rich SIP Trunking or migrate at their own speed With over 155 million active Microsoft Office 365 users, many companies are now considering adopting Microsoft Teams as their primary means of communication. Microsoft surveyed IT professionals and found that 41% expect their employers to move to Teams by the end of 2020.

nexVortex in Education – A Managed SIP Trunking Case Study

Telecommunication services don't have to take a back seat when budget constraints come to call Educational Think tank Manhattan Institute reports that America’s per-pupil spending on kindergarten through grade 12 education has nearly tripled in the last 50 years, and the country spends more money per student than any other major developed nation.

mSIP Trunking: Is it the Option Your Business Needs?

Managed SIP Trunking (mSIP) is a money-saving and scalable alternative to traditional service and it’s growing in popularity.   Perhaps you’ve decided that your Primary Rate Interface (PRI) service is way too expensive. Or maybe your service provider has decided to discontinue its legacy PRI offering. Maybe you’ve  already made the move to a SIP service but are looking for better performance and better guarantees for all your calls – including your IP to IP calls.