Deploy in LATAM Regions

We are able to deploy in LATAM regions to reduce latency and improve call quality for customers in Latin America.

Why regional deployment helps

Placing media and signaling close to callers reduces RTT and one‑way latency.
Shorter paths also reduce PDV/jitter and loss by avoiding congested or long‑haul links.

Private backbone routing

When traffic must cross regions, routing over the Synthflow backbone (instead of the public internet) can:

  • Reduce hop count and variable queuing delays.
  • Stabilize jitter by avoiding congested public segments.
  • Improve loss characteristics under peak loads.

Note
Actual results depend on last‑mile access and your carrier. Validate with the measurement steps in the case study below.

Regionalization playbook

  • Start with current call distribution and busiest geographies.
  • Measure baseline metrics (see performance metrics page).
  • Request a regional PoP deployment near the callers.
  • Re‑route calls/media to the new region or over the Synthflow backbone.
  • Re‑measure and compare before/after.

Case study — LATAM regionalization

Background
A major customer in LATAM experienced variable quality when traffic egressed via US infrastructure during peak hours.

Problem
Not only higher latency, but also packet loss and packet delay variation (PDV/jitter) led to choppy/robotic speech and occasional brief “squeal” during bursty delivery.

Intervention
We deployed a LATAM point of presence and routed inter‑region traffic over the Synthflow backbone instead of the public internet.

Results
Significant improvement in RTT (end‑to‑end) and overall call quality.

Before/After metrics

MetricBefore (US)After (LATAM)ΔNotes
RTT e2e (ms)191.399.5−48%Measured end‑to‑end
One‑way latency (ms)95.749.8−48%p50 / p95
PDV/jitter (ms)[TBD][TBD][TBD]%p95
Packet loss (%)[TBD][TBD][TBD]%p95
MOS[TBD][TBD][TBD]E‑Model
Call setup time (s)[TBD][TBD][TBD]%100→200 total

Measurement methodology

  • RTT/Path: mtr -u -c 200 <media-endpoint> and ping -c 200 <media-endpoint>.
  • RTP stats: Capture a sample of ≥100 calls with Wireshark; use RTP Stream analysis for jitter/loss/skew.
  • Synthetic traffic: sipp for call timing; optional iperf3 -u for UDP jitter/loss characterization when a test server is available.
  • MOS: Compute using E‑Model from measured one‑way delay, jitter, and loss.

What to watch for

  • ISP changes and peering shifts that re‑introduce jitter.
  • Packet bursts at hour boundaries (dialer starts) causing short‑term PDV spikes.
  • SBC/NAT keep‑alive intervals.

Replication checklist

  • Confirm call distribution to LATAM PoP.
  • Enable backbone routing for inter‑region segments.
  • Verify codecs (start with G.711).
  • Tune jitter buffer on PBX/SBC if configurable.
  • Re‑measure and compare p50/p95/p99 against baseline.

Metric definitions

MetricMeaning
RTT e2e (Round-Trip Time)How long it takes for your voice signal to go to the other person and back — like the “echo delay.”
One-way latencyThe delay from when you speak to when the other person hears you.
PDV / jitterHow much that delay changes over time — if it jumps around, voices can sound choppy.
Packet lossHow many bits of voice data get lost in transit — too much loss makes speech cut out.
MOS (Mean Opinion Score)A 1–5 quality score for how clear the call sounds overall.
Call setup timeHow long it takes after you dial before the call starts ringing or connects.