Why Call Quality Fails (and What to Do About It)
Side By Tech Engineering
When calls sound bad, the first instinct is to blame the phone or the headset. That is almost never the problem. Call quality issues are almost always network, configuration, or codec problems. Here is how to find the real cause.
The three real causes
1. Network jitter and packet loss
Voice over IP is real-time. Unlike downloading a file, you cannot retransmit a lost packet and play it a second later. When packets arrive out of order or drop entirely, you hear choppy audio, robotic voices, or one-way audio. A jitter buffer can smooth minor variation but it cannot compensate for a network that drops 3-5% of packets.
What to check: run a SIP-aware latency test, not a generic ping. Generic pings use ICMP which gets different priority treatment than UDP voice traffic on most networks.
2. Codec mismatch
Codecs compress voice audio. Different codecs have different bandwidth requirements and different quality levels. G.711 sounds great but uses 87 kbps. G.729 uses 32 kbps but sounds compressed. When a call negotiates a codec neither end prefers, you get unnecessary quality degradation.
What to check: review your PBX codec priority list. Most systems should prefer G.711 on local calls and OPUS or G.722 for anything that supports wideband.
3. QoS not configured
Quality of Service (QoS) marks voice packets so your router knows to prioritize them over file downloads and video streaming. Without QoS, a single employee downloading a large file can degrade every active call in the office.
What to check: most managed switches and enterprise routers support DSCP marking. If your voice VLAN traffic is not tagged, it is not being prioritized.
What is almost never the cause
- The physical phone or headset (unless it is visibly damaged)
- The number of users on the system
- The carrier (carriers rarely cause call quality problems - your local network does)
- The PBX software version (unless you just upgraded and it started then)
How to diagnose it
Start with a packet capture during a bad call. Tools like Wireshark let you filter on RTP traffic and see packet loss, jitter, and out-of-order delivery directly. This takes about 10 minutes and will tell you more than an hour of guessing.
If you do not have someone who can run a packet capture, the next best thing is to run the MOS (Mean Opinion Score) test from your SIP provider or PBX vendor. Most hosted PBX platforms include call quality reporting that surfaces jitter and packet loss metrics per call.
When to call us
If you have ruled out the obvious (QoS, codec settings, network path) and the problem persists, we can pull logs, run a proper network assessment, and identify the specific point of failure. We have seen every variation of this problem across offices ranging from 5 people to several hundred, in environments from basic cable internet to data center colocation.
Have a call quality problem?
Book a 15-minute call. We will ask the right questions and tell you where to look.
Book a Strategy CallIf your systems feel harder than they should be, they probably are.
15 minutes. No pressure. Just a real conversation about your systems and what is slowing you down.
AWS Certified · Cisco Certified · SIP School (SSCA) · 15+ years in production
