How does VoIP work from a network perspective (and why it sometimes doesn’t)? VoIP uses a unique set of network protocols – some for setting up calls, such as SIP (Session Initiated Protocol) and others for the actual call session, such as RTP (Real Time Protocol). VoIP success can be stymied if any of those protocols are not properly transmitted between the IP Telephones and the VoIP system Server and ultimately the recipient of the call.
During a call, VoIP call quality can suffer from three issues related to the network – packet loss, high latency, or high jitter (packet delivery disorder). All three of those conditions can arise for a number of reasons, but more commonly result from simple network congestion.
With that in mind, before you can make sure your network is ready for VoIP, you first need to understand what is on your network and how your network is working.
Things that you should look for at this point include:
-Applications that are most active in the network. Pay particular attention to those that are using large amounts of bandwidth on a continual basis, such as streaming music or video, big file transfers, and data backups.
-Areas of high utilization: Congestion is the enemy of voice quality, so it is important to recognize where VoIP might be choked out. Start with WAN links, where bandwidth is usually already bottlenecked, but don’t forget to assess the LAN as well, which can have congestion issues of its own.
-Presence of any existing IP-based voice or video traffic: you may have VoIP on your network and not even be aware of it. For instance, Skype and Google Voice are present on many networks _ particularly those that allow BYOD _ and Microsoft Lync is present as well since it is now a bundled option in MS Office suites as well as MS cloud and web-based offerings. Recognizing these products is important in understanding and planning the user side of the rollout, while also helping to reveal how the network is currently handling VoIP traffic.
One good approach to gauging VoIP readiness is to test the network using VoIP traffic generators from various points in the network. What is typically looked for is the quality of the test calls which is referred to as a MOS Score (Mean Opinion Score). The MOS Score takes into account: loss, latency, and jitter. A MOS score ranges from 1 to 5, where 5 is the best voice quality and 1 is the worst voice quality.
Quality of Service or QOS, is another term that is important. Optimizing the network for long term VoIP success will almost certainly require the use of QOS policies, which are used to mark specific types of traffic for high priority delivery by the network. Typically, VoIP is assigned a higher priority than almost anything else that happens on your network. This will result in less drops, less latency, and less jitter and make for better phone calls.