RTPProxy Reference

Role In Scenario

rtpproxy is the media relay for the pbx2 scenario. OpenSIPS controls it over the local control socket, and the service exposes a large public RTP range on the host.

The current setup is intentionally permissive and records every anchored session so the lab can demonstrate RTP bleed and flooding behavior while keeping captured media available through the scenario web surface.

• OpenSIPS controls rtpproxy on 127.0.0.1:7722/UDP
• public RTP range is 35000-40000/UDP
• advertised media address comes from PUBLIC_IPV4
• active recordings are written under /recordings/spool
• completed packet captures are written under /recordings in .pcap format
• pbx2 currently anchors media on IPv4 only

Key Files

File	Purpose
build/rtpproxy/run.sh	launches rtpproxy with the DVRTC port range and recording options
build/rtpproxy/healthcheck/healthcheck.sh	service health validation
build/opensips/config/opensips.cfg	rtpproxy control calls during offer/answer handling
compose/pbx2.yml	runtime wiring, recording volume, and host-network setup

Intentionally Vulnerable Behavior

• every anchored session is recorded unconditionally with -a
• the public RTP range is large and predictable enough for probing exercises
• arbitrary RTP toward active media targets can grow recordings rapidly during flood exercises
• the training setup remains permissive enough for RTP bleed-style probing

Verification

./scripts/compose.sh --scenario pbx2 run --rm testing dvrtc-checks rtp-bleed --host 127.0.0.1
./scripts/testing-run-all.sh --scenario pbx2
./scripts/compose.sh --scenario pbx2 logs rtpproxy

Related Documentation

• Overview
• Architecture
• OpenSIPS Configuration
• Nginx Reference