📡 Real-Time VoIP Monitoring, Built for Ops Teams
Calllytics gives you instant visibility into call quality and SIP health across your VoIP stack. Track MOS, jitter, packet loss, latency, and error rates — then alert your team before users notice.
Works with Asterisk, FreeSWITCH, Kamailio, SBCs, and SIP trunks.
Active calls
128Across 3 trunks
Avg MOS
4.2Good quality
Packet loss
0.6%Below threshold
SIP errors (5m)
3Mostly 408/503
Tip: set alerts by region, trunk, or endpoint group to reduce noise.
⚙️ How it works
Calllytics focuses on deterministic VoIP signals — not black-box decisions. It ingests SIP/RTP health data, calculates quality metrics, and triggers alerts when thresholds are crossed.
1) Collect
Capture registrations, call setup events, RTP stats, and node health.
2) Analyze
Compute MOS, jitter, packet loss, latency, call success rate, and SIP error trends.
3) Alert
Notify via email/Slack/webhooks when quality drops or failures spike.
Filter by trunk, region, site, or customer. Jump from overview → drilldown in seconds.
Track uptime and call success rate trends to avoid surprise incidents and churn.
Monitor quality without storing call content or transcripts.
📊 Key metrics
Everything you need to detect degraded call quality fast — with clean trends, per-trunk drilldowns, and alert rules that stay predictable.
MOS • jitter • packet loss • one-way latency • RTP stream health
Registrations • call setup time • 4xx/5xx rates • retries • timeouts
Call success rate • trunk availability • incident timeline • SLA reporting
Avg MOS < 3.8 for 3 minutes on trunk Carrier-A.
503 rate > 2× baseline in region EU-WEST.
Packet loss > 1.5% across endpoints in site Casablanca-HQ.
🔌 Integrations
Plug Calllytics into your existing VoIP environment. Start with one node, then scale to multiple regions.
Asterisk • FreeSWITCH • Kamailio • SBCs
Per-carrier dashboards and failover visibility
Prometheus / Grafana-friendly metrics
Email • Slack • Webhooks • on-call tools
Telecom teams need predictable operations. Calllytics uses transparent calculations and threshold-based rules so alerts stay explainable and consistent.
MOS/RTP statistics computed from standard VoIP signals.
Rule-based alerts reduce false positives and escalation fatigue.
Monitor quality without storing call audio or transcripts.
💳 Pricing
Choose a plan based on your monitored nodes and peak concurrent calls. No per-seat pricing.
Small VoIP environments
- ✓1 node monitored
- ✓Up to 500 concurrent calls
- ✓Core dashboards (MOS, jitter, loss, SIP errors)
- ✓Email alerts
- ✓7-day retention
Growing providers & NOC teams
- ✓Up to 5 nodes monitored
- ✓Up to 5,000 concurrent calls
- ✓Advanced drilldowns (by trunk / region / site)
- ✓Email, Slack & webhook alerts
- ✓SLA & availability reports
- ✓30-day retention
Telecom-scale operations
- ✓Unlimited nodes & regions
- ✓High-volume call monitoring
- ✓Custom metrics, thresholds & exports
- ✓Multi-tenant dashboards (optional)
- ✓Dedicated onboarding & support
- ✓Custom retention & compliance options
✉️ Contact
Tell us what you run (Asterisk/FreeSWITCH/Kamailio/SBC), your peak concurrent calls, and where issues show up. We’ll recommend the cleanest setup and alert thresholds for your environment.
No call content required. Monitor SIP/RTP signals only.
Every alert maps to a metric + threshold (no black box).
SLA visibility across nodes, trunks, and regions.
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