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VoIP call quality on SIP trunks over the internet

Will VoIP call quality be good over the internet on FirstNet SIP trunks?

Business Voice Minutes · Answered by FirstNet Technology Services

Short answer

Yes, on a suitable access circuit. When you also take FirstNet connectivity, FirstNet engineers quality of service (QoS) end to end so voice is prioritised. Best-effort internet works, but call quality over it cannot be guaranteed in MOS (Mean Opinion Score) terms.

In detail: VoIP call quality on SIP trunks over the internet

What determines call quality:

  • An uncontended link, ideally dedicated fibre, with about 100 Kbps per concurrent call plus 20% headroom.
  • One-way latency under 150 ms, jitter under 30 ms and packet loss under 1%.
  • Voice marked and prioritised (DSCP EF) from your LAN through to the provider.
  • Firewall settings, with SIP ALG disabled and RTP ports open.
  • Codecs, with G.711 as the baseline and G.729 or Opus where required.

FirstNet's own voice core runs on dual PortaOne clusters with zero-downtime updates, and calls reach other operators over direct interconnects. FirstNet's SLA covers what it manages: trunk availability, signalling, termination over its network and support.

For links FirstNet does not manage, it recommends the right access for your call profile and can run a 24-hour line monitor that measures jitter, latency and loss before you commit.

Source: FirstNet Business Voice Minutes service page →

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