Expert Advice on VoIP and SIP Services for Small Business / Technology / Last Modified: February 22, 2017

VoIP, SIP, bandwidth, and proxy servers ... what do they mean for your business calls? This interview with Nik Korchewsky explains it...

You have probably heard of Session Initiation Protocol, or SIP, but you may not be sure what it means. SIP is basically a signaling protocol that's used for controlling communications, like voice or video calls over the internet. SIP can be used for initiating, modifying, and terminating two-party or multi-party communications sessions, and it's closely entwined with Voice over Internet Protocol (VoIP) telephony. spoke with Nik Korchewsky of about VoIP and SIP services for small businesses. Q. On your website, you say that SIP allows individuals or businesses to easily move to better technologies as they emerge. Is SIP a part of a VoIP calling plan that you enroll in, or is it something separate or added on?

A. The growing thirst among communications providers,their partners and subscribers for a new generation of IP-based services is now being quenched by SIP -- the Session Initiation Protocol. An idea born in a computer science laboratory less than a decade ago, SIP is the first protocol to enable multi-user sessions regardless of media content and is now a specification of the International Engineering Task Force (IETF). Today, increasing numbers of carriers, CLECs and ITSPs are offering such SIP-based services as local and long distance telephony, presence & Instant Messaging, IPCentrex/Hosted PBX, voice messaging, push-to-talk, rich media conferencing, and more. Independent software vendors are creating new tools for developers to build SIP-based applications as well as SIP software for carriers' networks.

Network equipment vendors are developing hardware that supports SIP signaling and services. There is a wide variety of IP phones, user agents, network proxy servers, VOIP gateways, media servers and application servers that all utilize SIP.

Gradually, SIP is evolving into a powerful emerging standard. However, while SIP utilizes its own unique user agents and servers, it does not operate in a vacuum. Comparable to the converging of the multimedia services it supports, SIP works with a myriad of preexisting protocols governing authentication, location, voice quality, etc.

Clearly, SIP is an important protocol that is becoming widely deployed. SIP is a catalytic protocol that delivers key signaling elements, which can turn a voice over IP network into a true IP communications network -- a network capable of delivering next generation converged services. SIP is powerful, and yet simple. But that power comes from doing what it does best, and playing nicely with the rest to the other protocols in the converged protocol sandbox.

Flexible, extensible and open, SIP is galvanizing the power of the Internet and fixed and mobile IP networks to create a new generation of services. Able to complete networked messages from multiple PCs and phones, SIP establishes sessions much like the Internet from which it was modeled.

Q. Internet connection quality is very important in the utility of a VoIP phone system, and there are various ways a business can test its current internet connection to determine if it is adequate for the task. Can you elaborate a bit on what business owners should do to determine if their internet connection is good enough for high-quality VoIP calls?

A. It's not simple question. My recommendations are: apply QoS to prioritize VoIP traffic and make sure you have enough bandwidth on your Internet link. That's all that you need for quality VoIP service (and you must be sure about your VoIP provider quality).

The problem with IP is that, like Ethernet, it is a connection-less technology and does not guarantee bandwidth. Specifically, the protocol will not, in itself, differentiate network traffic based on the type of flow to ensure that the proper amount of bandwidth and prioritization level are defined for a particular type of application. By contrast, the cell-based ATM standard incorporates such service requirements in its specifications. Because IP does not inherently support the preferential treatment of data traffic, it's up to network managers and service providers to make their network components aware of applications and their various performance requirements.

You may wonder, but Internet Protocol IP-based networks provide "best effort"data delivery by default.

Best-effort IP allows the complexity to stay in the end-hosts, so the network can remain relatively simple. This scales well, as evidenced by the ability of the Internet to support its phenomenal growth. As more hosts are connected, network service demands eventually exceed capacity, but service is not denied. Instead it degrades gracefully. Although the resulting variability in delivery delays (jitter) and packet loss do not adversely affect typical Internet applications (email, file transfer and http/Web applications), other applications cannot adapt to inconsistent service levels. Delivery delays cause problems for applications with real-time requirements, such as those that deliver multimedia, including two-way applications like VoIP telephony services.

Quality of Service (QoS) generally encompasses bandwidth allocation, prioritization, and control over network latency for network applications. There are several ways to ensure QoS, no matter what type of network we are talking about --- Ethernet or ATM, IP or IPX.

The easiest one is simply to throw bandwidth at the problem until service quality becomes acceptable. This approach might involve upgrading the backbone to a high-speed technology such as Gigabit Ethernet. If you have fairly light traffic in general, more bandwidth may be all you need to ensure that applications receive the high priority and low latency they require.

However, this simplistic strategy collapses if a network is even moderately busy. In a complex environment --- one that has a lot of data packets moving in many paths throughout the network, or that has a mixture of data and real time applications --- you could run into bottlenecks and congestion.

Increasing bandwidth is a necessary step for accommodating these real-time applications, but it is still not enough to avoid jitter during traffic bursts. Even on a relatively unloaded IP network, delivery delays can vary enough to continue to adversely affect real-time applications. To provide adequate service --- some level of quantitative or qualitative determinism --- IP services must be supplemented. This requires adding some smarts to the net to distinguish traffic with strict timing requirements from those that can tolerate delay, jitter and loss. That is what Quality of Service (QoS) protocols are designed to do. The goal of QoS is to provide some level of predictability and control beyond the current IP "best-effort" service.

Simply adding bandwidth does not address the need to distinguish high-priority traffic flows from lower-priority ones. In other words, all traffic is treated the same. In the network realm, such egalitarianism is not good, because network traffic is, by its nature, unpredictable. For instance, on some days, you will see traffic bursts occurring at 8 a.m., while on other days you will see them at noon or at the end of the day. These traffic bursts can move around too. One day,  your Internet gateway or one of your switches is the bottleneck; another day, it's your intra-campus video conferences or heavy voice traffic causing the congestion.

Voice transmissions are real-time by the nature; hence the different approach to handling the network issues.

Packet loss, jitter and out-of-order packets are tied closely to each other. Taking care of a single network issue can often reduce all three problems and significantly improve the quality of voice calls.

We have pretty well explained typical VoIP problems at our web site:

Q. When companies shift from the older, PBX phone systems, they are concerned about maintaining their existing phone numbers for continuity of service to customers and clients. What should a company do to ensure they carry their existing numbers to a new VoIP system?

A. There are a lot of options how you can use your old infrastructure and get integration with latest VoIP platforms. You can use IP trunking interfaces or FSX / FXO gateways to bring the old analog phone line into a modern IP-PBX and your IP-based infrastructure.

Q. There's something called "SIP Enable" by Patton and Vitelity that allows analog or digital PBX users to switch to VoIP and save money while getting the longest useful life out of an existing PBX system. Do you think this is a worthwhile interim option, or should a company go ahead and make the switch to VoIP, since they'll have to replace their phone system eventually anyway?

A. I think VoIP is a must-have for sure, especially if a company's extensions are based on industry standard. SIP is a standard for VoIP industry nowadays, so if you have to choose between old-school hardware that has no option to expand and being used with VoIP, there is no sense to buy it. Actually I don't remember the last time I saw an offer for solution with hardware being analog PBX. Almost all PBX are IP-enabled nowadays.

Q. The concept of "failover" is important to a lot of businesses, particularly those that interact with customers frequently. What questions should a business ask of potential VoIP providers to learn whether re-routing or other "rescue" protocols will be sufficient to minimize any VoIP service interruptions?

A. I think that any service provider who respects its customers must have as much redundancy as possible. With VoIP service providers, failover termination trunks are must haves. Also, some advanced algorithm for balance loading and failover switch must be used. I think the best way to examine potential VoIP providers is to get a free trial account from them and make some test calls to the numbers/directions you plan to dial often. This way, you can make sure the voice quality is not just good, but perfect. Also see the average connection time and if there is a CLI (Caller ID) working well. Some routes could be "gray" and you can have bad quality calls even with "big name" VoIP service providers. You never know until you try, so my advice will be to try and see if it works for you and if it worth the price.

Photo Credit: Kevin Shorter

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