Quality of Service (QoS) can solve a lot of headaches that come with VoIP technology. For the end user, large delays are burdensome and can cause bad echoes. It’s hard to have a working conversation with too large delays. You keep interrupting each other. Jitter causes strange sound effects but can be handled to some degree with “jitter buffers” in the software. Packet loss causes interrupts. Some degree of packet loss won’t be noticeable, but lots of packet loss will make sound lousy. Let’s read more about QoS, how we measure it, and why it matters.
Why does QoS matter?
A phone call with bad audio quality feels terrible. The people on both sides of the line can feel frustrated and lose patience. No matter the conversation, if it’s hard to hear, you’d rather the call just drop.
Businesses have all kinds of conversations such as:
- Sales demos and webinars
- Customer support
- Employee interviews
- Team meetings with business leaders
All these conversations are essential. Inconsistent VoIP packets can lead to gaps in brand trust and communication failures. While we’ve all been annoyed by these issues, knowing how to fix them starts with learning how VoIP works. Setting up QoS improves the call quality for everyone on your Local Area Network (LAN). With CallHarbor, you don’t have to worry about any setup or maintenance of QoS. We have QoS monitors within our system to make sure your Quality of Service stays where it needs to be.
With remote work higher than ever, people are making business calls on the same network as YouTube, Netflix, and regular data traffic demands within the household. This has made prioritizing VoIP traffic more important than ever.
Not all VoIP service providers are equal. It’s relatively cheap for companies to set up a VoIP system, but that doesn’t mean it’s reliable. This has resulted in different VoIP providers offering vastly different levels of reliability and call quality. You always want to check the service level agreement and quality of service before committing to a provider. CallHarbor has a 99.999% SLA, and our quality of service is always monitored to assure you don’t encounter latency or jitter – and if you do, we can nip it in the bud.
- Latency measures the time it takes a packet of data to arrive at the destination IP address. All VoIP systems and all networks have some latency. Voice data packets are sensitive to delays exceeding 150 milliseconds each way.
- Network jitter measures the variation in packet delays, such as latency. When it comes to real-time voice, packets to arrive out of order with an unstable network connection. As a result, VoIP calls become unintelligible. Jitter above 30 ms will impact voice calls substantially.
- Packet loss measures the number of packets lost after transmission. Voice data packets are particularly sensitive to any packet loss. Anything above 3% packet loss means the audio quality will become significantly reduced.
- LAN and WAN network topologies determine whether you can influence its endpoints. Local Area Network (LAN) refers to the network managed by your router. Wide Area Network (WAN) is the broader network on the internet. VoIP packets travel from your phones through your LAN and on the WAN to reach their destination.
All these factors contribute to the audio quality of your VoIP calls. You don’t have to be an expert to set up VoIP QoS to improve your call quality, because you leave that to us!