Sample report

A test result people can understand and use.

Explore a representative SIPforge report and see how each section helps a team decide whether a voice system is ready.

SIPforge test report

Production readiness · SIP + RTP

Asterisk primary · 15 minute run · Generated 14 July 2026

Target met

Ready for the planned traffic profile

The server sustained 1,500 calls per minute with a 98.7% answer rate. Voice quality remained excellent and infrastructure utilization stayed below the warning threshold.

Call success

98.7%

Healthy

Voice quality

4.18

Excellent MOS

Avg. jitter

18 ms

Below 30 ms

Packet loss

0.4%

Below 1%

Call success over time

Target: 95% or higher

98.7% avg.

Peak server load

CPU64%
Memory58%
File descriptors41%

SIP response breakdown

200 OK98.7%
486 Busy0.8%
503 Unavailable0.5%

Observation

A small 503 spike appeared during the final load step. It did not affect the overall target, but should be monitored at higher traffic levels.

Scenario INVITE + RTPTransport UDPCodec G.711 µ-lawTotal calls 22,500

What each section tells you

From headline answer to technical evidence.

01

Executive summary

A plain-language conclusion states whether the system met the test goal and flags the most important risk.

02

Traffic and call outcomes

Total, successful, and failed calls show reliability at the exact call rate and concurrency tested.

03

Media quality

MOS, jitter, and packet loss show how the call sounded—not only whether SIP signaling succeeded.

04

Infrastructure behavior

CPU, memory, file descriptors, and Asterisk channels reveal whether capacity pressure caused degradation.

05

Failure evidence

SIP response codes and failure detail help engineering teams move directly from symptom to investigation.

06

Test configuration

Scenario, transport, codec, duration, and load parameters make the result repeatable and defensible.

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