"CI PBX" is a VoIP echo server, designed for CI/CD pipelines. Implemented in Go using diago.

Purpose

• Test SIP/RTP call flows in automated environments
• Validate call establishment and media transmission
• Check codec negotiation
• Provide reliable testing for continuous integration pipelines

Usage

# Basic usage (echo server only)
cipbx --listen 127.0.0.1 --port 5090

# Using short flags
cipbx -l 127.0.0.1 -p 5090

# Specify transport (defaults to udp). Supported: udp, tcp, tls, ws, wss
cipbx --transport udp -l 127.0.0.1 -p 5090

# With authentication (accepts REGISTER requests)
cipbx -l 127.0.0.1 -p 5090 -u username -w password

# Multiple accounts (comma separated lists)
cipbx -l 127.0.0.1 -p 5090 -u alice,bob -w secret1,secret2

# With timeout (automatically hang up after 30 seconds)
cipbx -l 127.0.0.1 -p 5090 -t 30

# With RTP payload validation (expect 0x30 in all payload bytes. Note: validate encoded value, not decoded)
cipbx -l 127.0.0.1 -p 5090 --expect 0x30

Features

• Echo Server: Call echo@<server-ip> to test echo functionality
• Playback Server: Call playback@<server-ip> for demo playback functionality
• Call Bridging: Call any other extension to bridge calls (e.g., alice@<server-ip>)
• Authentication: Optional digest authentication for REGISTER requests (supports multiple accounts) with 1-hour expiration
• Call Timeout: Optional automatic call termination after specified duration (in seconds)
• Transport Selection: Choose between udp, tcp, tls, ws, wss (default udp)
• RTP Payload Validation: Optional validation of RTP payload bytes for testing purposes

Call Routing

• echo@domain: Routes to echo server for testing media transmission
• playback@domain: Routes to playback server (demo functionality)
• user@domain: Bridges calls to the specified user (requires proper SIP routing)

Authentication

When username and password are provided:

• Accepts authenticated REGISTER requests
• Supports specifying multiple accounts using comma-separated lists (e.g. -u alice,bob -w secret1,secret2)
• Sets registration expiration to 1 hour
• Logs successful registrations

Example with authentication:

# Enable authentication for single account
cipbx -l 127.0.0.1 -p 5090 -u testuser -w testpass

# Enable authentication for multiple accounts
cipbx -l 127.0.0.1 -p 5090 -u alice,bob -w secret1,secret2

# Client can now register with:
# REGISTER sip:127.0.0.1:5090 SIP/2.0
# Authorization: Digest username="testuser", realm="cipbx", ...

Call Timeout

The timeout feature allows you to specify a maximum call duration in seconds. When enabled:

• Calls will be automatically terminated after the specified duration
• The server sends a BYE message to properly close the call
• Useful for testing scenarios where calls should not run indefinitely

Example with timeout:

# Set 60-second timeout
cipbx -l 127.0.0.1 -p 5090 -t 60

# Combine with authentication
cipbx -l 127.0.0.1 -p 5090 -u testuser -w testpass -t 120

RTP Payload Validation

The --expect flag enables RTP payload validation for testing purposes. When specified, the server will:

• Ignore the first 15 RTP packets (startup/comfort noise)
• Validate a window of payloads (≥50 packets over 3 seconds)
• Check that every payload byte matches the expected value
• Log validation results with RTP_ASSERT_OK or RTP_ASSERT_FAIL

Example usage:

# Expect 0x55 in all RTP payload bytes
cipbx -l 127.0.0.1 -p 5090 --expect 0x55

# Combine with other options
cipbx -l 127.0.0.1 -p 5090 -u testuser -w testpass --expect 0x55 -t 60

The validation is only active for echo@domain calls and logs results like:

• RTP_ASSERT_OK codec=PCMU bytes=0x55 (success)
• RTP_ASSERT_FAIL codec=PCMU expected=0x55 valid_packets=45 total_packets=50 (failure)

RTP tester

RTP testing utility for validating media transmission without SIP signaling.

🎯 Purpose

• Test raw RTP media streams directly
• Validate codec performance (PCMA, PCMU, Opus)
• Debug RTP packet transmission issues
• Measure media quality without SIP overhead

🔄 Difference from cipbx

While cipbx tests complete SIP/RTP call flows, rtptester focuses exclusively on the media layer:

• rtptester: Raw RTP testing only (no SIP)
• cipbx: Complete SIP + RTP call testing

Basic Usage

# With custom values using flags
rtptester --local-ip 192.168.1.100 --local-port 6000 --remote-ip 192.168.1.200 --remote-port 6001 --codec PCMU --debug

# Using short flags
rtptester -l 192.168.1.100 -p 6000 -r 192.168.1.200 -P 6001 -c PCMU -d

Manual Testing with GStreamer

The RTP tester can be manually tested using GStreamer to send and receive RTP packets. This is useful for verifying the echo functionality works correctly.

Test Setup

Run these commands in separate terminals:

Terminal 1 - RTP Echo Tester:

.\rtptester --local-ip 127.0.0.1 --local-port 5004 --remote-ip 127.0.0.1 --remote-port 5005 --codec PCMA --debug

Terminal 2 - GStreamer Sender:

gst-launch-1.0 audiotestsrc wave=sine freq=440 num-buffers=300 ! audioconvert ! audioresample ! alawenc ! rtppcmapay ! udpsink host=127.0.0.1 port=5004

Terminal 3 - GStreamer Receiver:

gst-launch-1.0 udpsrc port=5005 ! application/x-rtp,encoding-name=PCMA,clock-rate=8000 ! rtppcmadepay ! alawdec ! audioconvert ! wavenc ! filesink location=received_audio.wav

Expected Results

• The RTP tester will log received and echoed packets
• The received_audio.wav file will contain the echoed audio
• You should hear a 440Hz sine wave when playing the audio file

Direct GStreamer Test (Bypass RTP Tester)

To verify GStreamer setup works independently:

Terminal 1 - Direct Sender:

gst-launch-1.0 audiotestsrc wave=sine freq=440 num-buffers=300 ! audioconvert ! audioresample ! alawenc ! rtppcmapay ! udpsink host=127.0.0.1 port=5005

Terminal 2 - Direct Receiver:

gst-launch-1.0 udpsrc port=5005 ! application/x-rtp,encoding-name=PCMA,clock-rate=8000 ! rtppcmadepay ! alawdec ! audioconvert ! wavenc ! filesink location=direct_test.wav

This should create direct_test.wav with audio content, confirming GStreamer is working correctly.