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.
Turn complex voice agent load testing into a clear workflow with SIPforge. Simulate realistic traffic and catch connectivity bottlenecks before your clients do.
No SIPp scripting Real-time 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
A centralized SIP load testing tool for the voice systems your team relies on
- 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
One SIP load testing tool for clear test insights.
One clean dashboard to see what happened, view the call matrix, and prove server stability under load.
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 voice load testing your system handles.
Hear quality in the numbers
Understand MOS, jitter, and packet loss at a glance.
Evidence on demand
Turn the same test into a report anyone can review.
From test setup to sign-off
Voice load testing that delivers clear answers.
View capacity, call quality, and failure patterns together, so your team can move from “it should work” to “we proved it works”.
Live call visibility
Monitor call matrix, 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
Measure performance changes across test runs and make informed decisions about SIP enhancement.
Reports built for decisions
Export reports in PDF, CSV, or JSON to give clients or stakeholders evidence they can use.
How it works
SIP load testing in three simple steps.
The platform handles SIP testing and metric collection, allowing your team to stay focused on what the result means.
Try it nowConnect your voice stack
Add a SIP server, a Twilio account, or an AI voice agent. Pre-flight checks catch setup issues before a run.
Model real traffic
Configure your scenario, call rate, concurrency, duration, transport, and codec—without writing SIPp XML.
See what breaks first
Monitor calls, debug SIP call failures, and server health. Compare runs and share a decision-ready report.
From a readiness run
The conclusion, ready to repeat.
Explore anonymized examples of how a representative SIP and RTP test report helps teams evaluate call performance and investigate infrastructure issues.
“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. QA and engineering could review the same test evidence.”
Ready when you are
Make your next voice release a confident one.
Create your workspace and run SIP load testing in minutes.