The days of traditional phone lines, even ISDN PRI, are fading. Cloud-based systems and computer networks offer more adaptable, feature-rich, and cost-effective voice communication solutions. To tap into these advantages, adopting packet-based phone service, also known as VoIP, is key.
People’s feelings about VoIP as a replacement for traditional phone service are mixed, ranging from enthusiastic support to staunch opposition. This spectrum of opinions stems from individual experiences with the technology and when they first used it.
Today’s leading VoIP providers offer a vastly different experience compared to the early days when VoIP was primarily seen as a way to cut down on long-distance charges. While connecting a coder/decoder (CODEC) to broadband internet could turn an analog phone into an IP phone, essentially transforming it into a computer peripheral able to share the network, early implementations often suffered from poor call quality.
The primary culprit behind this subpar voice quality was the design of most computer network equipment, which heavily relied on the TCP/IP protocol. While this protocol is excellent for internet data transfer as it diligently resends any lost data packets until the transmission is complete, it’s not ideal for real-time communication like voice calls. With TCP/IP, the priority is accurate delivery, not speed, so delays are expected.
VoIP, unlike file transfers, demands real-time data transmission and is therefore highly sensitive to network hiccups. Even a tiny delay of half a second, negligible for data transfer, can disrupt the flow of a VoIP call, leading to awkward pauses and a “walkie-talkie” effect.
Adding to the challenge, lost data packets, while easily resent in TCP/IP, can’t be retrieved in real-time VoIP communication. Since requesting a resend would further disrupt the conversation flow, VoIP protocols simply ignore these missing packets, resulting in distorted audio.
Even when packets arrive successfully, variations in their travel time, known as jitter, can negatively impact call quality. While some jitter can be mitigated with buffering, excessive jitter leads to the same latency issues as packet loss and can also scramble the order of arriving packets, further degrading the audio quality.
Voice calls are inherently more susceptible to network issues than one-way communication methods. To guarantee high-quality and reliable VoIP calls, several factors are critical.
Top-tier VoIP providers circumvent many of these problems by hosting calls on their own private networks, bypassing the public internet. These “SIP trunks,” named after the Session Initiation Protocol crucial for VoIP signaling, provide dedicated connections between a business’s phones and the provider’s switching equipment, allowing for strict control over latency, jitter, and packet loss.
The choice of CODEC also significantly influences voice quality. Wideband CODECs can deliver excellent call clarity, and High Definition (HD) CODECs offer even more natural and intelligible audio. Opting for narrowband CODECs to accommodate more calls on a single SIP trunk may compromise call quality. The choice depends on the desired call quality.
If you’re considering VoIP to replace your existing phone system but have concerns about call quality, researching reputable high-reliability VoIP service providers is crucial. These providers offer a range of features and benefits to ensure a smooth transition and superior call experience.
