What a voice bandwidth estimate can tell you

Enough bandwidth creates room for calls. It does not guarantee that audio arrives on time.

Count the packets, not just the codec

This model uses 20 ms of audio per packet: 50 packets per second. Basic RTP and UDP headers add 12 and 8 bytes. A base IPv4 header adds 20 bytes; IPv6 adds 40. The per-direction rate is the codec payload rate plus header bytes × 8 ÷ packetization milliseconds.

For G.711 at 64 kbps, this gives 80 kbps per call per direction over IPv4, or 88 kbps over IPv6. Multiply by concurrent calls and the selected added margin. A 25% added margin means multiply by 1.25; it does not mean 25% of the final link capacity is held free.

Leave room in both directions

A duplex audio call sends and receives. Compare the displayed upload requirement with the upload capacity allocated to voice, and the download requirement with the download allocation. Do not compare a doubled sum against each direction. Other applications, busy-hour traffic and the actual media path still matter.

Know what is excluded

Ethernet and Wi-Fi framing, VLANs, VPNs, SRTP tags, RTP extensions, RTCP, SIP signaling, redundancy, retransmission, video and unrelated traffic are outside the base model. Opus targets are planning inputs for an adaptive codec, not fixed observed rates. The adjustable margin is not a universal standard or a guarantee that every excluded cost fits.

Choppy or robotic audio

Compare wired Ethernet with Wi-Fi where practical. Pause large transfers and inspect packet-loss and jitter statistics during an affected call. Collect the time and scope of the issue. A speed test alone cannot explain uneven packet arrival or burst loss.

One-way or missing audio

Check mute and selected input/output devices first. If repeatable, give your administrator the call time, direction and affected endpoints so they can inspect media routing, NAT and the specific firewall policy. Do not disable the firewall or broadly open ports as a generic fix.

Delay or echo

For long pauses, ask the administrator to review congestion and the approved media/VPN path. Ping is round-trip timing, not measured one-way audio latency. For echo, try a headset, lower speaker volume and avoid two active audio devices in the same room. These are isolation steps, not automatic diagnoses.

Sources: Cisco voice bandwidth calculations, RTP header structure, Opus over RTP, Microsoft media-network guidance.