There are a lot of reasons why VoIP phone system users can experience poor call quality. The good news is that many of these issues are correctable. Also, many issues are just temporary. VoIp-Info.org has listed many of these issues and recommends how to solve them.

Jitter is a frequent complaint from VoIP users. Jitter is found on packet switched networks. Packets often arrive at their destination in a different order than intended. The result is scrambled audio quality. One solution to jitter is a jitter buffer. It stores packets to reduce delay variations.

Another possible problem with VoIP calls is latency. VoIP latency is similar to an echo, which occurs to do transmission delays. There are three kinds of delays impacting VoIP calls. They include: propagation delay, handling delay and queuing delay. Prioritizing VoIP network traffic, however, can eliminate both latency and jitter. Ways to prioritize VoIP traffic include policy-based network management and bandwidth reservation, as well as changes to type of service, class of service, and multi-protocol label switching. Selecting the right VoIP router can correct many of these concerns.

Of course, a poor Internet connection can greatly impact VoIP call quality. A poor router can further compound these issues. In the same vein, improperly configured internal networks can be a prime culprit of low quality VoIP. Naturally, using a properly configured, business-class, high-speed internet service with compatible VoIP routers is the solution.

There are other factors outside of network configuration that can also affect VoIP performance, each with their own unique solution. For instance, as described by VoIP Mechanic, to fix acoustic echo, lower the volume of the phone. For electromagnetic interference, move the IAD, or base, of a cordless phone. Also, take away splitters or separate caller ID devices between the IAD and the phone.  Users may want to use a higher-quality phone cord, or a corded phone instead of a cordless phone. Lower-quality echo cancellers at the IP-PBX (News - Alert) or at the destination speaker can also cause echo, VoIP Mechanic said.   

VoIP callers may also experience choppy voice on a connection. This may be caused by insufficient bandwidth, VoIP Mechanic said. The website suggests turning off other computers to see if the choppy voice continues. Also, conduct a bandwidth test both for upload and download, the report said. Also, look for malware, spyware, and sharing software, which often eat up available bandwidth

Still, other VoIP phone quality issues relate to buzzing, interference and static. VoIP Mechanic explains that they are caused by a signal or voltage in the VoIP connection that causes an unwanted sound. One solution is to disconnect everything from the phone port of the ATA (analog telephone adapter).  Then connect one corded phone to the ATA.  If there are weak signals between the base station and handset there may be too long of a distance or drained batteries in the handset.  Another possibility could be interference from another wireless device.

In related news, GL Communications (News - Alert), a provider of test, analysis and simulation products and services, has announced new solutions to improve testing of such issues.

"Voice quality testing within VoIP and PSTN networks is growing, a major concern is testing the VoIP phones and the analog telephone adaptors (ATA) that will be used within the VoIP/PSTN networks,” Vijay Kulkarni, the company’s CEO, said in a recent statement. “The VoIP phones and ATAs must be free of noise and other hindrances that may cause a degraded voice quality when using this equipment. The GL VQuad software combined with GL's Voice Quality Testing (VQT) software permits this desired VQT testing, of the VoIP equipment, yielding results that include PESQ/PAMS/PSQM+ Mean Opinion Scores (MOS). These results provide an excellent overall measure of voice quality of the equipment under test.”

Also, GL’s Dual Universal Telephony Adapter (Dual UTA) tests the voice quality of telecommunication networks.

“Used in conjunction with GL's VQuad software application, the Dual UTA simplifies end-to-end testing of voice quality, echo, noise, and other impairments of TDM, VoIP, and Wireless networks,” Kulkarni added. “A GPS sync port permits precise synchronization of sending and receiving audio functions, one-way delay (OWD), and round trip delay (RTD). The Dual UTA can be directed to automatically send and record sample voice files between nodes of a telephony network. These files can then be submitted to GL's Voice Quality Testing (VQT) software application for analysis according to widely accepted International Telecommunications Union (ITU) voice comparison algorithms. These algorithms include PESQ, PAMS and PSQM.”