Product Roadmap

See what CallMeter offers today and what is coming next. SIP stress testing, continuous monitoring, 150+ real-time metrics, and much more.

CallMeter is built to be the most complete SIP and VoIP testing platform available. This roadmap shows what you can use today, what is coming soon, and what we are planning for the future.

Feature Requests Welcome

Have a feature you need? Contact us at support@callmeter.io or through the in-app support portal. Customer requests directly influence our development priorities.

Available Now

These features are live and available on the platform today.

Generate hundreds to thousands of concurrent SIP endpoints with configurable audio and video media. Simulate real-world call patterns with staggered buildup timing, multiple groups, and cross-group targeting. Each endpoint registers with your SIP infrastructure, places or receives calls, exchanges real RTP media, and collects quality metrics throughout the call lifecycle.

Configure every aspect of the test: number of endpoints, call duration, registration timing, codec selection, media type, and call flow. Whether you are validating a new SIP trunk, load-testing a PBX, or benchmarking a carrier switch, CallMeter generates the traffic patterns you need at the scale you need.

Continuous Monitoring with Probes

Deploy automated quality monitoring that runs around the clock. Probes place SIP calls at configurable intervals (every 5, 15, 30, or 60 minutes) and evaluate quality thresholds after each execution. When quality degrades, probes transition through health states (HEALTHY, DEGRADED, UNHEALTHY, UNKNOWN) and trigger webhook notifications so your team can respond before users are affected.

Each probe runs as a lightweight test with all the same metric collection capabilities as a full stress test. Define thresholds on any metric — MOS score, jitter, packet loss, registration time — and let CallMeter watch your infrastructure continuously. See Creating a Probe for setup instructions.

150+ Real-Time Metrics

Over 150 measurements collected per endpoint per second during every call. Metrics span seven categories:

Quality — MOS (Mean Opinion Score), R-Factor, jitter, round-trip time, packet loss percentage, and clock analysis.
Network — Packet counts, bitrate (send and receive), sequence number analysis, and duplicate detection.
RTCP Feedback — NACK counts, PLI (Picture Loss Indication), FIR (Full Intra Request), and SLI (Slice Loss Indication).
Jitter Buffer — Buffer delay, late packets, buffer overruns, retransmission requests, and adaptive buffer behavior.
Audio — Packet Loss Concealment events, audio level monitoring, silence detection, comfort noise, and Opus-specific diagnostics.
Video — Frame rate (sent and received), freeze events, freeze duration, keyframe intervals, resolution tracking, and decode metrics.
Call Timing — Post-Dial Delay, Answer Seizure Ratio, Network Effectiveness Ratio, call setup time, and registration duration.

Every metric is collected independently for both the send and receive direction. See the Metrics Glossary for the complete list.

Multi-Codec Support

Full audio support for four codecs: PCMA (G.711 A-law), PCMU (G.711 mu-law), G.722 (wideband), and Opus (adaptive bitrate). Video support for three codecs: H.264 (Constrained Baseline profile), VP8, and VP9.

Select codecs per group to match your production configuration exactly. SDP offer/answer negotiation determines the final codec selection, just as it would in a real call. See Supported Codecs for detailed specifications.

Cloud and Self-Hosted Workers

Two worker deployment models to cover every testing scenario:

Cloud Workers — Platform-managed workers available in multiple regions. No setup required.
Self-Hosted Workers — Deploy workers on your own infrastructure for testing internal SIP systems behind firewalls, inside private networks, or at specific geographic locations.

Both worker types use the same metric collection, the same status lifecycle, and the same management interface. Mix cloud and self-hosted workers within a single test for hybrid testing scenarios.

Group-Based Testing

Create tests with multiple groups, each with fully independent configuration: registrar, codecs, media, worker region, role (caller/receiver), and endpoint count. Enable cross-group calling to simulate realistic scenarios: Group A calls Group B across different registrars. See Groups.

Caller and Receiver Modes

Configure groups as callers (initiating SIP INVITE), receivers (waiting for incoming calls), or both. Cross-group targeting uses phased dispatch — receivers register and become ready before callers start dialing. Both sides report metrics independently. See Groups.

Mid-Call Scenario Actions

Define what happens during a call beyond basic dial-and-hangup. Actions are configured as When/Then cards with three trigger types: At Time, On Event, and On Event + Delay. Available actions: Hold, Resume, Start Sending Track, Negotiate Track, Update Media, and Send DTMF. Up to 30 scenario actions per group. See Scenario Actions.

DTMF and IVR Testing

Send and detect DTMF digits during active calls using three transport methods: RFC 4733 (RTP telephone-event), SIP INFO (application/dtmf-relay), or both simultaneously. Build IVR test scenarios with event-driven When/Then triggers that react to incoming DTMF digits. See Scenario Actions.

QoS / DSCP Marking

Per-track DSCP priority marking with RFC 4594 presets: Voice (EF/46), Video (AF41/34), Signaling (CS3/24), Low Priority (CS1/8), or custom values (0-63). See Advanced Track Settings.

Deferred Track Negotiation

Control when media tracks are negotiated and activated. Tracks can start with port 0 (negotiate mid-call) or start inactive with RTP keepalive (activate mid-call). Simulates video escalation, content sharing, and progressive media scenarios. See Advanced Track Settings.

SIP Session Timers (RFC 4028)

Automatic session refresh with configurable interval, Min-SE negotiation, and refresher role selection (UAC/UAS). Handles 422 responses automatically. See Session Timers.

100rel / PRACK (RFC 3262)

Reliable provisional response support. Send and acknowledge 1xx responses reliably for enterprise SIP environments that require PRACK.

DNS-Based Server Discovery (RFC 3263)

Automatic transport selection via NAPTR, SRV, and A/AAAA DNS lookups. Runtime resolution on each worker for correct behavior in split-DNS environments. See Transport Protocols.

Outbound Proxy Support

Route all SIP signaling through a proxy server via RFC 3261 Route header. Test SBC traversal, corporate SIP proxies, and multi-hop routing. See Outbound Proxy.

RTCP-MUX (RFC 5761)

Single-port RTP and RTCP multiplexing to reduce port usage and simplify NAT traversal. Includes RFC 8858 holdconn maintenance during hold. See RTCP Multiplexing.

SDES-SRTP Encryption (RFC 4568)

Media encryption via SDP key exchange with three cipher suites (AES-128-HMAC-SHA1-80, AES-128-HMAC-SHA1-32, AEAD-AES-128-GCM). Configurable policy: Disabled, Offered, or Required. See SDES-SRTP.

SIP over TLS

Encrypted SIP signaling with PEM certificate upload, X.509 validation, mutual TLS support, custom CA, and automatic sips: URI scheme. TLS 1.2 and 1.3 support. See TLS Certificates.

DTLS-SRTP Encryption (RFC 5764)

WebRTC-grade media encryption via DTLS handshake with auto-generated ECDSA P-256 certificates. Configurable SRTP method bitmask: SDES-only, DTLS-only, or dual offer for maximum compatibility. See DTLS-SRTP.

SIP Transport Protocols

Full support for SIP signaling over UDP, TCP, TLS, and WSS (WebSocket Secure). See Transport Protocols.

Custom Media Files

Upload your own audio and video files for realistic media playback during tests. Automatic transcoding into all supported codec formats at upload time. See Uploading Files.

Role-Based Access Control

Five-level role hierarchy (Owner, Admin, Editor, Tester, Viewer) with per-project scoping. See Roles and Permissions.

REST API

Programmatic access for CI/CD integration and automation. See API Endpoints and CI/CD Integration.

Public Status Pages

Real-time SLA dashboards powered by probe health data. See Status Pages.

Test Run Visualization

Group topology diagrams, grouped endpoint grids, paired endpoint navigation, test run timeline with phase bars, and per-endpoint lifecycle timelines with SIP event milestones.

Historical Analytics

Full per-second metric granularity stored for every test run. Browse history, compare runs, and identify quality trends over time.

Coming Soon

These features are in active development.

Network Impairment Simulation

Inject controlled network degradation at the worker level to test how your SIP infrastructure behaves under adverse conditions. Configure packet loss percentages, jitter profiles, latency injection, and bandwidth constraints per group. Simulate real-world conditions like congested Wi-Fi, satellite links, mobile networks, or cross-continental routing without modifying any network equipment.

AVPF with NACK and RTX

Real-time retransmission support using AVPF (Audio-Visual Profile with Feedback). Send and process NACK requests for lost packets and handle RTX (retransmission) streams. Test how your infrastructure handles feedback-based error recovery and measure the impact on video quality and latency.

ICE, STUN, and TURN

Full ICE (RFC 5245) support for NAT traversal testing. Workers gather STUN candidates, allocate TURN relays, and perform ICE connectivity checks. Test your STUN/TURN infrastructure under load and verify NAT traversal behavior in complex network topologies with multiple NAT layers.

Customer feedback comes first. Every feature request is tracked and weighted by the number of customers requesting it.
Enterprise customers can influence priority directly. If your team depends on a specific capability, talk to us.
Quality over speed. We ship features when they meet our quality bar.
Transparency is ongoing. This page is updated as features ship and new development begins.

Have a feature request or want to discuss prioritization? Reach out at support@callmeter.io or through the in-app support portal.

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Why CallMeter — How CallMeter compares to alternatives
Platform Architecture — How the system works

Product Roadmap

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