chitholian/voip-call-simulator repository overview

VoIP Call Simulator
A minimal, containerized Asterisk 22 instance that places and receives SIP calls with human-like, randomized behavior — built for load-testing and validating call-switching / routing software against a CSV call-detail-record (CDR).

⁠What it does

Runs headless in two roles (set via MODE):

• callee – answers inbound SIP calls with a realistic state machine: post-dial delay → ring → early media → answer → talk → hangup, each phase governed by tunable probabilities (no-answer, busy, decline, hangup) and lognormal/exponential durations.
• caller – fires outbound calls at a target peer via Asterisk AMI Originate, with random caller-ids/destinations, ramp/concurrency control, a per-second rate limit, and randomized ring/answer/cancel timing.

Every answered, busy, unanswered, or failed call is written to a CSV CDR on both sides, so you can reconcile your switching software's behavior against the recorded disposition.

⁠Quick start (callee – answer calls)

docker run -d --name voip-callee --network host \
  -e MODE=callee \
  -e SIP_PORT=5060 \
  -e AMI_PORT=5038 \
  -e AMI_USER=sim -e AMI_SECRET=callsim \
  chitholian/voip-call-simulator:latest

The container listens on UDP/TCP 5060 (PJSIP) and answers inbound calls via a state-machine AGI. Dial sip:6888@<host>:5060 from any UAC/softphone.

⁠Quick start (caller – make calls against a peer)

Point CALLER_PEER_HOST/CALLER_PEER_PORT at a callee instance or your target SIP server, then:

docker run -d --name voip-caller --network host \
  -e MODE=caller \
  -e SIP_PORT=2280 \
  -e AMI_PORT=5039 \
  -e AMI_USER=sim -e AMI_SECRET=callsim \
  -e CALLER_PEER_HOST=127.0.0.1 -e CALLER_PEER_PORT=5060 \
  -e CALLER_FROM_PREFIXES=9101,9202 -e CALLER_FROM_LEN=4 \
  -e CALLER_TO_PREFIXES=6888 -e CALLER_TO_LEN=4 \
  -e CALLER_MAX_TOTAL_CALLS=500 -e CALLER_MAX_CONCURRENT=50 -e CALLER_RATE_PER_SEC=10 \
  chitholian/voip-call-simulator:latest

The caller will ramp to 50 concurrent calls at 10/sec, stop after 500 originate attempts, and exit with code 0 once calls drain.

⁠Configuration

All knobs are environment variables. Key ones:

See the repository README.md for the full variable reference and implementation notes.

⁠CDR output

CSV call records are written to /var/log/asterisk/cdr-csv/Master.csv inside the container (omit --rm or copy it out):

docker exec voip-callee cat /var/log/asterisk/cdr-csv/Master.csv

Includes dispositions for answered, busy, no-answer, and failed calls (unanswered=yes), with UTC timestamps, so you can reconcile against the target software.

⁠Build & publish

• Local build: docker compose up -d --build (alias docker build -t voip-asterisk ./asterisk).
• Automatic publish: pushes to GitHub Container Registry (ghcr.io) on every push to main, tagged latest and with the commit SHA.
• Base image: andrius/asterisk:22 (Asterisk 22).

⁠License

Open source. See the repository for details and to report issues.