VoIP has its distinct advantages and disadvantages. The greatest advantage of VoIP is price and the greatest disadvantage is call quality. For businesses who deploy VoIP phone networks -- particularly those who operate busy call centers (customer service, tech support, telemarketing, et cetera) -- call quality issues are both inevitable and unacceptable. To analyze and fix call quality issues, most of these businesses use a technique called VoIP call monitoring.
VoIP call monitoring, also known as quality monitoring (QM), uses hardware and software solutions to test, analyze and rate the overall quality of calls made over a VoIP phone network [source: ManageEngine]. Call monitoring is a key component of a business's overall quality of service (QoS) plan.
Call monitoring hardware and software uses various mathematical algorithms to measure the quality of a VoIP call and generate a score. The most common score is called the mean opinion score (MOS). The MOS is measured on a scale of one to five, although 4.4 is technically the highest score possible on a VoIP network [source: TestYourVoIP.com]. An MOS of 3.5 or above is considered a "good call" [source: ManageEngine].
To come up with the MOS, call monitoring hardware and software analyzes several different call quality parameters, the most common being:
- Latency -- This is the time delay between two ends of a VoIP phone conversation. It can be measured either one-way or round trip. Round-trip latency contributes to the "talk-over effect" experienced during bad VoIP calls, where people end up talking over each other because they think the other person has stopped speaking. A round-trip latency of over 300 millisecond is considered poor [source: TestYourVoIP.com].
- Jitter -- Jitter is latency caused by packets arriving late or in the wrong order [source: SearchVoIP.com]. Most VoIP networks try to get rid of jitter with something called a jitter buffer that collects packets in small groups, puts them in the right order and delivers them to the end user all at once. VoIP callers will notice a jitter of 50 msec or greater [source: TestYourVoIP.com].
- Packet loss -- Part of the problem with a jitter buffer is that sometimes it gets overloaded and late-arriving packets get "dropped" or lost [source: TestYourVoIP.com]. Sometimes the packets will get lost sporadically throughout a conversation (random loss) and sometimes whole sentences will get dropped (bursty loss) [source: TestYourVoIP.com]. Packet loss is measured as a percentage of lost packets to received packets.
There are two different types of call monitoring: active and passive. Active (or subjective) call monitoring happens before a company deploys its VoIP network. Active monitoring is often done by equipment manufacturers and network specialists who use a company's VoIP network exclusively for testing purposes [source: VoIP Troubleshooter.com]. Active testing can't occur once a VoIP network is deployed and employees are already using the system.
Passive call monitoring analyzes VoIP calls in real-time while they're being made by actual users [source: VoIP Troubleshooter.com]. Passive call monitoring can detect network traffic problems, buffer overloads and other glitches that network administrators can fix in network down time.
Another method for call monitoring is recording VoIP phone calls for later analysis. This type of analysis is limited, however, to what can be heard during the call, not what's happening on the actual network. This type of monitoring is usually done by human beings, not computers, and is called quality assurance.
On the next page, we'll talk about making VoIP calls using cell phones.