Built for every team that runs voice or video over SIP
What you can’t measure precisely,you can’t improve.
Know your SIP infrastructure’s quality to the second. Prove it to anyone who asks.
Generate thousands of concurrent SIP calls. Probe every trunk and interconnect continuously. Measure 90+ quality metrics per call, per second, live. See every issue the moment it happens — and prove what your infrastructure delivers to anyone who needs the evidence.
- Measure precisely — 90+ live quality metrics per call, per second. Signaling, media, and network in one view.
- Watch continuously — probes across every trunk, route, and carrier interconnect. Catch quality drops before customers do.
- Prove independently — public status pages, signed reports, and audit-ready exports. Evidence that holds up.
- Scale realistically — 10 to 10,000+ concurrent calls with real codecs and network conditions. No hardware, no local toolchain.
Web-based. First test in under 5 minutes. Deploy on-premise when traffic can’t leave your network.
The Trust Layer for SIP Infrastructure
90+
Metrics Per Call
24/7
Continuous Probes
Live
Status Pages
10K+
Concurrent Calls
How It Works
From setup to insights in three steps.
1.Configure
Point it at your SIP server. Set call volume, codecs, and thresholds. First test in under 5 minutes.
2.Test
Hit run. Thousands of real calls with real codec traffic hit your infrastructure — G.711, Opus, H.264, VP8, VP9. The same stream your users would hear and see.
3.Analyze
See exactly where quality degrades — which endpoint, which second, which direction. Export the evidence for your change management board.
From a number to the cause — every second, per endpoint, per direction.
90+ metrics per call — voice, video, packet, and timing. Per endpoint, per direction, every second.
RTP Quality
Packets, loss, jitter, sequence analysis, out-of-order, duplicates
Audio Quality
R-factor, audio level, VAD, comfort noise, Opus metrics, MOS (G.107)
Video Quality
Bitrate, FPS, resolution, freeze/artifact count/duration, keyframe tracking
Call Timing
PDD, setup time, ring time, time-to-first-media, call result
Continuous SIP Monitoring with Probes
Schedule automated test calls that run 24/7. Know about degradation before your customers do.
- Two modes: Active calling (CALLER) + Passive listening (CALLEE)
- Flexible scheduling: every 5 minutes or cron expressions
- Threshold monitoring across all quality metrics — set thresholds on any metric you choose
- Intelligent status: HEALTHY → DEGRADED → UNHEALTHY with sliding window
- HMAC-SHA256 signed webhooks with retry logic
- Auto-disable after repeated failures
SLA Enforcement & Public Status Pages
Token-shared public pages. Embeddable uptime badges. No login required.
Give your customers real-time visibility into SIP infrastructure health. No login required.
- Public status pages with token-based sharing (no login required)
- 24h rolling uptime percentage per probe
- Timeline of recent checks — color-coded health history
- Multi-probe dashboards for carrier/platform monitoring
- Flexible probe limits that scale with your needs
- Embeddable status badges for your website
Every call, every second, every direction.
Per-endpoint quality on the same timeline as the SIP ladder. Drilldown that holds up in a post-mortem — from 10 calls to 10,000+. One dashboard, one timeline, one source of evidence.
Book a DemoCloud workers or on-premise agents. One dashboard.
Use our managed cloud workers or deploy lightweight agents in your own data centers. Everything connects to one dashboard.
- Deploy workers on-premise — test SIP and RTP traffic stays in your network; only metric summaries ship to CallMeter
- One dashboard regardless of where workers run — one console for cloud and on-premise
- Workers authenticate with revocable tokens — revoke access instantly if compromised
- Capacity-aware dispatch — tests automatically route to workers with available capacity
Comprehensive Protocol Support
Full SDP negotiation, multiple RTP profiles, and every codec your infrastructure needs.
Audio Codecs
Video Codecs
SIP Methods
RTP/RTCP
Security
Quality Standards
Telephony Features
Owned SIP + RTP stacks. Measurement before smoothing.
Metrics come from the wire — before the jitter buffer smooths them and before the codec compensates for loss. MOS computed per ITU-T G.107. What the dashboard shows is what the endpoint actually received.
Dual independent collectors
Send-side and receive-side RTP measured separately — per SSRC, per direction, per second. No single-agent grading conflict. Follow RFC 3550 §A.3 for session-scoped metric handling.
Measurement at the wire
Jitter buffers smooth spikes and codecs compensate for loss. CallMeter extracts metrics before any of that happens. What the dashboard shows is what the endpoint actually received.
Video measured at audio depth
Same measurement depth for video as for audio. Millisecond-precision freeze detection, codec-specific quality analysis, frame-level diagnostics for H.264, H.265, VP8, and VP9.
Full Ownership Compounds
We control every layer of the SIP and RTP stack. That means we ship metrics others can’t capture, support protocols others can’t reach, and evolve as fast as the standards do.
Measurement depth, not signaling proxies.
CallMeter owns the full SIP and RTP stack. Every metric comes from the wire — before the jitter buffer smooths it and before the codec compensates for loss. Send and receive directions measured independently, per SSRC, per second. What the dashboard shows is what the endpoint actually received.
Book a DemoMeasure. Watch. Prove. Scale.
Evidence your team, your customers, and your contracts can trust — from measurement to proof.
Root Cause Investigation
When quality drops, correlate SIP signaling and per-SSRC RTP metrics on the same timeline. Know exactly where quality broke — which endpoint, which second, which direction.
Voice & Video Quality
90+ metrics per call, per direction, per second. Audio and video measured at the same depth — codec, jitter buffer, freeze detection, frame loss, MOS per G.107.
Continuous Monitoring
24/7 probes verify quality against your SLA thresholds across every trunk and interconnect. Public status pages share the evidence. Alerts fire before customers notice.
Independent Proof
Generate shareable quality reports from neutral infrastructure. Settle disputes, back SLA discussions, and show customers your quality with evidence they can verify.
SIP Load Testing
Run 10,000 concurrent calls with real codecs and network conditions. First test in under 5 minutes. See exactly where your infrastructure breaks — and why.
Network Impairment
Inject packet loss, jitter, latency, and duplication on live test calls. Validate how your infrastructure behaves under real-world network conditions before production does.
Your Data Center
Deploy test agents on-premise. SIP and RTP test traffic stays in your network; only metric summaries ship back to CallMeter.
API & CI/CD
REST API and CLI. Automate quality gates in your pipeline. Block the deploy that degrades call quality before it reaches customers.
Frequently Asked Questions
Common questions about SIP infrastructure testing, quality assurance, and how CallMeter works.
What is SIP testing and why do carriers need it?
SIP testing validates that your voice and video infrastructure performs under real-world conditions — not just that calls connect, but that quality holds at scale. Carriers need it because production failures cost revenue and trust. CallMeter surfaces your infrastructure's quality ceiling before customers do — 90+ quality metrics per call per second, continuous probes across every trunk and interconnect, and independent evidence you can share with partners.
Learn about capacity planning →How do you diagnose VoIP quality issues automatically?
CallMeter correlates SIP signaling events with per-SSRC RTP quality metrics automatically. When quality drops, you see the exact endpoint, direction, and second it degraded — along with the root cause (codec negotiation failure, packet loss spike, jitter threshold breach). No manual trace correlation needed.
Learn about incident diagnosis →Can I test network impairment effects on SIP call quality?
Yes. CallMeter simulates jitter, latency, packet loss, and packet duplication on live test calls. You see exactly how each impairment affects MOS, jitter buffer behavior, and call completion rates at any scale — before deploying to production.
Explore all features →How do you settle quality disputes between carriers?
CallMeter runs from independent cloud infrastructure — neither side owns the measurement. Both parties receive the same timestamped, tamper-resistant report. One test, one report, one truth. Disputes that took weeks of back-and-forth resolve same day.
Learn about dispute settlement →How does CallMeter verify SLA compliance independently?
Continuous probes run from infrastructure independent of both you and your provider. Quality is verified against your SLA thresholds 24/7 with automatic breach detection and timestamped evidence for credit claims. All providers in one dashboard.
Learn about SLA enforcement →Can I prove my SIP infrastructure quality to customers?
CallMeter generates a verifiable quality score based on real test data across audio and video. Embed a verification badge on your website, include it in RFP responses, and link to live proof. Be the first provider in your market to back quality claims with independent evidence.
Learn about quality proof →How is CallMeter different from open-source SIP testing tools?
Open-source tools test signaling only — they tell you if calls connect, not if quality holds. CallMeter generates and analyzes real RTP media, measures 90+ quality metrics per endpoint per second, and correlates signaling with media on the same timeline. Purpose-built for production-grade quality assurance, not just connectivity checks.
See the technology →How is CallMeter different from commercial SIP testing platforms?
Commercial SIP testing platforms typically sit on softphone libraries or CLI frameworks — measurement happens after the jitter buffer smooths it and after the codec compensates for loss. CallMeter owns the full SIP and RTP stack end-to-end: metrics come from the wire, send and receive directions are measured independently per SSRC (no single-agent grading conflict), video is measured at the same depth as audio, and MOS is computed per ITU-T G.107. One platform for measurement, continuous observation, and independent proof.
See the technology →Can MSPs and carriers white-label SIP quality assurance?
Yes. White-label CallMeter under your brand and offer quality assurance as a billable service to enterprise customers. Multi-tenant management, branded reports, deployment verification — new revenue line without years of internal development.
Learn about white-label →Prove what you deliver.
Second by second.
90+ quality metrics, continuous probes, and independent evidence — one platform.