TOLL FREE 888-808-6111
6 Factors that Contribute to VoIP Call Quality
Remember the time before flat screen high-definition (HD) TVs? Television sets were big and boxy, since they had to house all the innards of curvy cathode ray tube displays (the inspiration for the YouTube name), famous for their scan lines and relatively low resolutions. Although CRTs don't have fixed screen resolutions like LCDs, most only output a maximum picture of 480p, compared to the 720p, 1080p and 4K now. Content originally optimized for them is typically letterboxed when viewed on a widescreen TV.
Hosted VoIP solutions are to traditional private branch exchanges (PBXes) as HD TVs are to CRT TVs – a major upgrade in quality, yet backward-compatible with the most essential functions. Just as a modern TV can play video in 480p in addition to much crisper resolutions, a VoIP system can process voice calls while also supporting more advanced services such as distributed denial-of-service protections, auto attendants and softphone clients.
At the same time, VoIP services and the unified communications (UC) platforms they are often parts of vary significantly in both usability and reliability. Comparing their spec sheets can seem like evaluating the displays on HD TVs – they might seem very similar on the surface, but the fine-grained details can make all the difference in whether you get the experience you expect.
There are many attributes affecting VoIP call quality in particular, and you will want to ensure that your selection accounts for all of them. Let's look at six in more depth:
While VoIP at its best can seem like a simple way to communicate, it's underpinned by a complex mix of networking infrastructure that has to operate seamlessly in tandem. Routers, modems and firewalls are all integral components of modern VoIP, and their ages and build qualities can directly determine call clarity.
Suitable VoIP routers should have several key features, including:
The performance requirements of hosted VoIP necessitate network connections that don't get squeezed during peak usage. Many broadband plans don't fit the bill. They typically have much faster upstream than downstream speeds and, more importantly, are shared by multiple end users in the local footprint to control costs. Bandwidth can come under sufficient pressure to jeopardize VoIP call quality.
A practical solution is to invest in Dedicated Internet Access (DIA) from a trusted hosted VoIP provider like Telesystem. DIA is highly reliable, scalable and customizable, with adjustable increments and guaranteed bidirectional bandwidth not available from mass-market internet packages.
VoIP dervies many of its advantages from being delivered via packet-switch IP networks instead of circuit-switched telephone lines. This setup is why VoIP is much less costly for long-distance and also a lot easier to configure for administrators. However, it comes with a few drawbacks that require constant attention, such as jItter.
Jitter refers to the out-of-sequence arrival of packets. In VoIP calling, jitter can sharply degrade quality, since packets need to arrive sequentially in real-time for acceptable clarity The use of jitter buffers can help preserve the best possible experience. These mechanisms sit between incoming packets and the voice decoder, to collect packets and assemble them in the proper order.
Latency is a measure of delay. With VoIP, the most common signs of latency are echo and unintentional talking at the same time due to delays in the transmission of data. Delivery times for VoIP calls are measured in milliseconds, meaning that there's not much margin for error.
A delay might stem from issues in propagation over a network, forwarding of frames or packet queueing. Regardless, the result is the same: odd-sounding speech that distracts from the content of a call. The solution is to use traffic prioritization via the VoIP-optimized routers we mentioned. Leveraging MPLS links and incorporating multiple types of connectivity into a software-defined WAN (SD-WAN) is a further step to minimize latency and jitter for superior quality.
Packet loss refers to failures in packets reaching their intended destinations. It's a major cause of jitter and a common drag on VoIP call quality. If even 1 percent of packets are lost, callers will notice the resulting effects, such as audio briefly cutting out or sounding garbled.
There are multiple possible triggers for packet loss, including an overburdened network, hardware failures and the types of routing inefficiencies that SD-WANs can help correct. Implementing DIA, up-to-date equipment and SD-WAN services can greatly reduce the risks and effects of packet loss.
Codecs are at the heart of VoIP. They compress audio signals for transmission over an IP network and then decompress them for replay. Not all codecs deliver identical performance, nor do they have the same requirements for bandwidth and licensing.
A handful of codecs dominate VoIP telephony, including G.729, G.711, Speex and several others. Check with your hosted VoIP provider to see which codecs they utilize and if those standards are acceptable for your specific requirements.
Protocols, namely SIP and H.323, can also influence quality. SIP has superseded H.323 in most VoIP deployments, although the latter still has a presence in some video conferencing systems. SIP is generally easier to debug and much less resource intensive, allowing for more consistent call quality
Ready to get started with a hosted VoIP solution that works for your organization? Contact the Telesystem team today to learn more about how we can deliver the voice, DIA and SD-WAN services you need for the best possible communications.