Join Our Newsletter to Get Recent Industry Solution Updates | Subscribe
SEO preview (Phase 1). This is a clone for internal testing. The live voice-quality test page is unchanged. Do not share this URL publicly.

CCaaS voice quality test

Free browser test of contact-center voice quality and network readiness: MOS, latency, jitter, packet loss, UDP 3478, NAT and VPN against Genesys Cloud, Five9, NICE CXone, Microsoft Teams, Zoom, Google Meet and RingCentral. No download. Results stay on your device.

All tests run in your browser — no results are collected or stored.

Genesys Cloud regions to probe
Ready. The full test takes about 60–90 seconds and uses roughly 30–60 MB of data.

Diagnostics — share these with support if you need help

Recommendations

    Need help with these results?

    Star Telecom helps businesses design, deploy and optimize cloud contact center platforms — including the network readiness work these results point to. Get a free consultation →

    © Star Telecom · startelecom.ca · All tests run in your browser — no results are collected or stored.
    How this test works 
    Methodology:
• Latency/jitter: 24 timed HTTPS probes per target after 2 warm-ups (so TCP/TLS
  setup is excluded): Google edge (gstatic.com), Microsoft edge
  (login.microsoftonline.com — anycast on the Microsoft Global Network;
  teams.microsoft.com as automatic fallback), Zoom (zoom.us), Genesys
  Cloud regional APIs
  (api.cac1.pure.cloud / api.mypurecloud.com / api.usw2.pure.cloud), Five9
  (app.ca.five9.com Montréal, app.five9.com US), NICE CXone
  (cxone.niceincontact.com), RingCentral (app.ringcentral.com). Jitter = mean
  absolute difference between consecutive RTTs.
• Loss estimate: share of probes per target that fail or exceed 2.5 s — a proxy;
  browsers can't send raw UDP pings.
• UDP checks: WebRTC ICE against stun.l.google.com:19302 (Meet's transport) and
  stun.cloudflare.com:3478 (tests outbound UDP 3478 — required by Genesys Cloud
  and standard SIP/STUN). If blocked, media falls back to TCP and quality drops.
• Bandwidth: download/upload via speed.cloudflare.com (CORS-enabled), with
  adaptive upload chunk sizing over two parallel streams. Download counts are
  corrected to on-the-wire bytes via Resource Timing (encodedBodySize), since
  browsers report decompressed bytes and compressible payloads would otherwise
  overstate throughput several-fold; the result notes the endpoint used.
• NAT type: one WebRTC gathering pass against two STUN servers; the same local
  port mapping to different public ports = symmetric NAT (endpoint-dependent).
• Latency under load (bufferbloat): idle RTT baseline vs RTT sampled every
  ~150 ms while the speed tests saturate the link.
• Network path: public IP seen by STUN (media path) vs HTTPS egress IP;
  a mismatch indicates VPN/proxy split-tunneling.
• VPN/proxy assessment (heuristic): split-tunnel detection, egress AS
  organization classified against datacenter/VPN networks and corporate SASE
  gateways (Zscaler, Netskope, Prisma Access, Umbrella, Cloudflare ZT, …),
  and IP-geo longitude vs device UTC offset (>3.5 h apart = mismatch). A
  full-tunnel VPN on a normal ISP with matching geography is NOT detectable.
• Diagnostics: public IP / ISP / ASN / IP-based location from
  speed.cloudflare.com/meta (fallbacks: ipwho.is, api.ipify.org); IPv6 via
  ipv6.icanhazip.com; browser/OS/device from the user agent. Everything is
  shown locally in the browser only — nothing is uploaded or stored.
• MOS: simplified ITU-T E-model per target. Effective latency = one-way delay +
  2×jitter + 10 ms; R-factor −2.5 pts per 1% loss; standard R→MOS curve.

Platform requirements referenced:
• Google Meet: audio 12–100 kbps; 720p ~1.7 Mbps; HD group ~3.6 Mbps;
  outbound UDP 3478 & 19302–19309.
• Genesys Cloud: <150 ms one-way latency, <1% loss; Opus 32–128 kbps
  bidirectional per call; UDP 3478 (STUN) + UDP 16384–65535 (SRTP).
• Five9: <150 ms one-way latency to DC; ~128 kbps per agent (Agent Desktop
  Plus); RTP UDP 35000–65000; DCs: Santa Clara/Atlanta (US), Montréal (CA).
• Microsoft Teams: audio 58–76 kbps; <100 ms RTT to MS edge, <30 ms jitter,
  <1% loss; media UDP 3478–3481.
• Zoom: audio ~60–80 kbps; media UDP 8801–8810 (TCP 443 fallback).
• NICE CXone: WebRTC softphone over HTTPS 443, SRTP on UDP 6000–65531;
  <150 ms one-way latency.
• RingCentral: <150 ms one-way, <30 ms jitter, <1% loss; RTP UDP 20000–64999.

    CCaaS network readiness

    What this CCaaS voice quality test measures

    This is a network and VoIP quality test for agent workstations, not a call-recording QA or speech-analytics tool. Run it from the desk, home office, or BPO site you actually take calls from.

    A normal internet speed test only reports TCP throughput. Contact-center voice fails for other reasons: UDP blocked on the firewall, jitter on Wi-Fi, bufferbloat when someone uploads a file, a VPN splitting media away from HTTPS, or a distant Genesys / Five9 region. This browser test scores those conditions against the platforms agents actually use.

    MOS per platform

    Mean Opinion Score from a simplified ITU-T E-model using latency, jitter, and loss.

    Latency and jitter

    Probes Google Meet, Microsoft Teams, Zoom, Genesys Cloud, Five9, NICE CXone, and RingCentral.

    UDP 19302 and 3478

    STUN checks. If these are blocked, WebRTC falls back to TCP and audio quality drops.

    Bandwidth

    Download and upload with compression-corrected throughput.

    Bufferbloat

    How much latency rises when the line is busy.

    NAT, VPN, IPv6

    NAT type, split-tunnel / SASE heuristics, public IP, ISP, and IPv6.

    Nothing is uploaded or stored. Diagnostics stay on this device so you can copy or save a report for IT or a carrier.

    Results

    How to read your results

    Use MOS as the headline, then check UDP, jitter, and latency under load before you blame the CCaaS vendor.

    MetricGoodInvestigate
    MOS4.0 or higher (4.3+ is excellent)Below 4.0: callers will notice chop, delay, or missing syllables
    One-way latencyUnder 150 ms (Genesys, Five9, NICE, RingCentral)Over 150 ms one-way / 300 ms RTT
    Microsoft Teams RTTUnder 100 ms to the Microsoft edgeOver 100 ms RTT, or jitter over 30 ms
    JitterUnder 30 msChoppy or robotic audio, especially on Wi-Fi
    Loss estimateUnder 1%Dropped words; often congestion or a bad radio path
    UDP 3478 / 19302ReachableBlocked: media will TCP-relay or fail
    Latency under loadUnder ~30–100 ms extraLarge jump = bufferbloat; enable SQM/QoS on the router

    Platforms

    Platform requirements

    The test probes the same class of edges your softphone uses. Thresholds below are the vendor targets the scoring references.

    Genesys Cloud CX network test

    Aim for under 150 ms one-way latency and under 1% loss. Opus typically uses 32–128 kbps bidirectional per call. Open outbound UDP 3478 (STUN) and UDP 16384–65535 (SRTP). Probe Canada (cac1, Montréal), US East, and US West before an official WebRTC Diagnostics run.

    Five9 connectivity test (browser)

    Five9 wants under 150 ms one-way to the data center and about 128 kbps per Agent Desktop Plus session. Media uses RTP over UDP 35000–65000. This is the no-download check you can send a remote agent in 90 seconds. Montréal and US endpoints included.

    NICE CXone network test

    The integrated WebRTC softphone signals over HTTPS 443 with SRTP on UDP 6000–65531. Target under 150 ms one-way, under 30 ms jitter, and under 1% loss.

    Microsoft Teams

    Audio is roughly 58–76 kbps. Microsoft’s media targets are under 100 ms RTT to its network edge, under 30 ms jitter, and under 1% loss. Media prefers UDP 3478–3481.

    Zoom, Google Meet, and RingCentral

    Zoom audio is about 60–80 kbps on UDP 8801–8810. Google Meet needs UDP 3478 and 19302–19309. RingCentral targets under 150 ms one-way, under 30 ms jitter, under 1% loss, RTP on UDP 20000–64999.

    How the test works

    24 HTTPS probes per target after warm-ups; jitter from consecutive RTTs; loss as probes that fail or exceed 2.5 s; WebRTC ICE against Google STUN 19302 and Cloudflare STUN 3478; MOS from a simplified E-model.

    Why this is not a speed test

    Why a VoIP speed test is not enough

    Pages that rank for “VoIP speed test” often measure bulk download/upload. Contact-center voice is a real-time UDP problem.

    You can have 500 Mbps and still fail Genesys Cloud if UDP 3478 is blocked, if Zscaler/Netskope hairpins media, or if bufferbloat adds 200 ms whenever a laptop backups to the cloud. This CCaaS network readiness test is built for that gap.

    Troubleshooting

    Common reasons CCaaS voice fails

    • Wi-Fi instead of wired Ethernet, or a congested 2.4 GHz radio (high jitter)
    • Corporate VPN or SASE (Zscaler, Netskope, Prisma Access, Umbrella) without split-tunnel for media
    • Firewall missing UDP 3478 / platform media ranges
    • Symmetric NAT forcing TURN relays and one-way audio
    • Bufferbloat on the CPE when other devices saturate the link
    • Agent registered to a far-away Genesys or Five9 region

    If the report points to SIP, BYOC, or carrier path rather than the LAN, Star Telecom’s BYOC SIP connector and Canadian SIP trunking are the usual next step. For platform design and operations, see Genesys Cloud CX and support and managed services.

    Related tools: the telecom cost calculator and the Genesys Cloud CX AI token calculator.

    CCaaS voice quality

    Frequently asked questions

    Answers for contact-center IT and Genesys Cloud teams running a browser network-readiness test.

    What is a CCaaS voice quality test?
    It is a browser check of whether your network can carry real-time contact-center voice (MOS, latency, jitter, loss, UDP) to platforms such as Genesys Cloud, Five9, and NICE CXone. It is not a QA score of recorded calls or an AI speech-analytics product.
    How is this different from a VoIP speed test?
    A VoIP speed test usually emphasizes bandwidth. This test also probes each platform’s edge, UDP/STUN, NAT, bufferbloat, and VPN split-tunneling, then returns a MOS per platform.
    How is this different from Genesys WebRTC Diagnostics or Five9 CAT?
    Those are official, vendor-specific tools (Genesys often needs a login; Five9 CAT typically needs a download). Use this page as a no-install pre-check, then run the vendor tool if you need their support to accept the evidence.
    What MOS score is good enough for a contact center?
    Treat 4.0 as the floor for professional voice and 4.3 as the comfortable target. Below 3.6, plan a network change before you add more concurrent agents.
    Why does UDP 3478 matter for Genesys Cloud?
    Genesys Cloud WebRTC uses STUN on UDP 3478 and SRTP media on high UDP ports. If 3478 is blocked, setup fails or media relays over TCP and quality collapses.
    Can I run this for remote or WFH agents?
    Yes. Send them this page and ask for the saved report. Run it on the same Wi-Fi or Ethernet they use for the softphone, with the VPN in the state they use on shift.
    Does Star Telecom store my results or IP?
    No. The test runs in your browser. Copy or save the report locally if you need to send it to support.
    Which Genesys Cloud regions can I probe?
    Canada (cac1, Montréal), US East (mypurecloud.com), and US West (usw2). Enable the ones you actually use.
    Will a VPN or Zscaler/Netskope break voice quality?
    Often. Secure web gateways add latency and drop UDP. Exempt voice media with split tunneling, then re-run this test to compare.
    How long does the test take and how much data does it use?
    About 60–90 seconds and roughly 30–60 MB, mostly from the bandwidth portion.

    Need a hand

    Need help with these results?

    Star Telecom helps businesses design, deploy and optimize cloud contact center platforms — including the network readiness work these results point to.

    Get a free consultation

    Savings Calculator

    Please fill your information in fields below

    * This field is mandatory
    * This field is mandatory
    * This field is mandatory
    * This field is mandatory

    Yes I would like to receive email from Star Telecom about the latest in technology news, white papers, webcasts, events and more. Please check to subscribe. You may opt out at any time

    PRIVACY STATEMENT: Please note we will never sell, rent or loan your email address to any outside company or organization.