There are three critical performance issues that need to be considered prior to VoIP deployment:
1)
Latency --- the end-to-end delay is delivering the voice stream from the speaker's mouth to the listener's ear.
2)
Jitter --- the unpredictable, variable delays in the delivery of each voice packet.
3)
Packet Loss -- the dropping of individual packets caused by network congestion. Each of these three issues can cause significant degradation in voice quality and overall system reliability.
Because VoIP is real-time, two-way communication, it is very sensitive to delays in the network. Acceptable VoIP quality requires a latency or delay of not more than 80ms each way for true toll-quality voice communication. Voice quality degrades as latency increases, but with a delay of 150-180ms, voice quality is still in the "acceptable" range. In addition to the voice stream itself, latency issues must also be addressed with other VoIP protocols (SIP, H.323, MGCP, etc.) that handle the call control functions between two systems or users. In fact, these signaling protocols are often even more sensitive to delays in the network.
Because IP is a "best effort" protocol, if left unattended it will always be subject to unpredictable performance, including packet loss. Like jitter and latency, packet loss can be very disruptive to VoIP's performance. This is usually not an issue in the corporate LAN, but at the bandwidth-constrained LAN/WAN boundary, where there is much greater contention for space in a smaller pipe, congestion and packet loss can become a serious problem. Although packet loss of one percent or less is within the bounds of toll quality voice, once packet loss reaches three percent or more the listener will notice the conversation "breaking up." Unless this problem is controlled, packet loss can lead to dropped calls and the possibility of VoIP system failure.
Latency is commonly associated with network congestion and variable router queue depth at the WAN router. Again, this is seldom a problem within the corporate LAN, where significant bandwidth is generally available to everyone who needs it. The real problem area is at the LAN/WAN boundary, as traffic transitions to smaller links. Some network devices attempt to overcome this problem by employing various queuing techniques to ensure voice packets take priority over other traffic waiting to get on the network. This is helpful to a certain extent, but being first in line to get on a crowded freeway doesn't mean you'll get to your destination quickly.
The more effective traffic management solutions that can classify all VoIP-related protocols, employ advanced rate control techniques to allocate a guaranteed amount of bandwidth to each traffic type, while preventing queues to form on router eliminating a primary cause of latency and jitter. These advanced capabilities are available today in the more advanced Application Traffic Management systems.
As noted above, jitter causes irregularities in the flow and delivery of data. This can be disruptive to a real-time application like VoIP. The tolerance range for VoIP jitter is in the range of 20-30ms. If jitter causes delays in excess of that --- especially on a consistent basis --- voice quality will suffer. Some VoIP vendors have tried to solve this problem by introducing their own jitter buffers or queues to temporarily store and "smooth out" the delivery of voice packets. Likewise, routers also offer queuing mechanisms for the same purpose. Both options, however, can exacerbate the problem by actually contributing to delays.
The Key to Controlling VoIP
The ability to control and minimize network jitter is a key capability of an Application Traffic Management system. The more advanced systems apply innovative technologies - TCP Rate Control (for data traffic) and UDP Rate Control (for voice traffic) - to assign and maintain a guaranteed rate and quality of all voice and data traffic across the WAN. These technologies are superior to conventional queuing, in that they "smooth" delivery of the traffic, but do so in way that reduces packet loss and retransmissions.
The key is to apply more controls to the IP network, moving it from "best effort" to predictable, optimal performance for all business-critical applications --- including VoIP. A capable Application Traffic Management system can deliver an unmatched level of traffic control and performance optimization at the LAN/WAN boundary. The value is delivered through a set of application visibility and control features that minimize the congestion and unpredictability of IP and maximize application performance over the existing WAN, often without the need for costly bandwidth upgrades.
