Realistic traffic scenarios
Run signaling, RTP media, registration, DTMF, or OPTIONS tests with guided controls.
Voice testing, made visible
Know your voice system can handle the call.
SIPforge turns complex voice load testing into a clear workflow—from realistic traffic to live quality metrics and client-ready reports.
No SIPp scripting Live test visibility Shareable evidence
Launch readiness test
INVITE + RTP · 1,500 CPM
Success rate
98.7%
Voice quality
4.18
Avg. jitter
18 ms
Peak calls
126
Call success over time
Stable at target load
Ready for the planned traffic level
Works with the voice systems your team already runs
- Asterisk
- FreeSWITCH
- Kamailio
- Twilio Voice
- AI voice agents
- Call success at target load
- 98.7%
- Voice quality, MOS
- 4.18
- Calls per minute sustained
- 1,500
Anonymized outcomes from a representative readiness run. See the sample report
See SIPforge in action
The full story of a test, on one screen.
A simple workspace shows what happened, how the calls sounded, and whether the server stayed healthy.
Actual product preview
Acme Voice LabsTests
All systems operationalHealthyAM
Test dashboard
Launch readiness test
CompletedINVITE + RTP · 1,500 CPM · UDP
View logsCompare run
Call Rate
25c/s
target reached
Active Calls
126
concurrent
Voice Quality
4.2
Excellent
Jitter
18ms
under 30 ms
Packet Loss
0.4%
under 1%
Success Rate
98.7%
Healthy
Traffic
Calls per second & voice quality
Calls/sec MOS quality
Target server
Server resources
CPU
64%
RAM
58%
Open FDs
428
Channels
126
Within safe range
No resource bottleneck detected.
See capacity
Know the traffic level your voice system handled.
Hear quality in the numbers
Understand MOS, jitter, and packet loss at a glance.
Leave with evidence
Turn the same test into a report anyone can review.
From test setup to sign-off
Clear answers at every stage.
See capacity, call quality, and failure patterns together—so your team can move from “it should work” to evidence.
Live call visibility
Follow call rate, active calls, success rate, and SIP response codes while the test is running.
Voice quality you can explain
Track MOS, jitter, and packet loss with plain-language thresholds for good and poor performance.
Server health in context
Correlate voice behavior with CPU, memory, file descriptors, channels, and call drops.
Run-to-run comparison
See whether a release, configuration, or capacity change improved or degraded the result.
Reports built for decisions
Export PDF, CSV, or JSON results and give clients or stakeholders evidence they can use.
How it works
From signup to a useful answer in three steps.
The platform handles test orchestration and metric collection. Your team stays focused on what the result means.
See every capabilityConnect your voice stack
Add a SIP server, Twilio account, or AI voice agent. Pre-flight checks catch setup issues before a run.
Model real traffic
Choose a proven scenario, call rate, concurrency, duration, transport, and codec—without writing SIPp XML.
See what breaks first
Watch calls, voice quality, failures, and server resources live. Compare runs and share a decision-ready report.
From a readiness run
The conclusion, ready to repeat.
Anonymized examples of what a representative SIP and RTP report makes easy to say.
“Sustained the planned call rate with voice quality still in the excellent range.”
“The failure spike sat next to CPU, so the capacity question had an answer.”
“Sign-off used the PDF conclusion. Engineering kept the SIP detail underneath.”
Ready when you are
Make your next voice release a confident one.
Create your workspace and run a meaningful test in minutes.