Getting Started with SipCLI — A Lightweight SIP ToolIntroduction
SipCLI is a compact, command-line-based Session Initiation Protocol (SIP) client designed for developers, system administrators, and power users who need a lightweight tool for testing, automating, and troubleshooting SIP-based VoIP systems. Unlike full-featured graphical softphones, SipCLI focuses on simplicity, scriptability, and reliable low-level interaction with SIP endpoints and SIP servers (registrars, proxies, and B2BUAs). This guide walks you through installation, basic usage, common tasks, scripting examples, troubleshooting, and practical tips to integrate SipCLI into your VoIP workflow.
What SipCLI is good for
SipCLI excels in scenarios where you need direct control and reproducibility:
- SIP registration and authentication testing
- Calling from the command line (INVITE/ACK/BYE flows)
- Automated dialing and call-quality checks
- SIP messaging (MESSAGE, OPTIONS) for presence and diagnostics
- Scripting call scenarios for load or functional testing
- Inspecting SIP headers and responses for debugging
Installation
Prebuilt packages
Many Linux distributions and third-party repositories provide prebuilt binaries for SipCLI. Check your distribution’s package manager first:
- On Debian/Ubuntu:
- sudo apt update
- sudo apt install sipcli
- On Fedora:
- sudo dnf install sipcli
- On Arch:
- pacman -S sipcli
If a package isn’t available, use the source-install method below.
Build from source
Requirements:
- A C/C++ compiler (gcc/clang)
- make and cmake (if project uses cmake)
- libssl (for TLS)
- libsrtp or equivalent (if SRTP support is needed)
- Optional: pcap/wireshark libraries for capture features
Typical build steps:
git clone https://example.org/sipcli.git cd sipcli mkdir build && cd build cmake .. make sudo make install Adjust commands to match SipCLI’s actual repository and build system.
Basic Concepts and SIP Primer
SIP is a text-based protocol used to establish, modify, and terminate multimedia sessions. Key SIP methods you’ll encounter with SipCLI:
- INVITE — start a call
- ACK — confirm a successful response to an INVITE
- BYE — terminate a call
- REGISTER — register an endpoint with a SIP registrar
- OPTIONS — query capabilities of a UAS
- MESSAGE — send instant messages
SIP works with SDP (Session Description Protocol) inside message bodies to negotiate media parameters (codecs, IP/port for RTP). SipCLI often exposes flags to supply SDP or use built-in media handling to generate/accept RTP streams.
Quick-start examples
1) Register to a SIP server
A minimal REGISTER command:
sipcli register --server sip.example.com --user alice --password s3cr3t --realm example.com Expected result: a 200 OK if credentials and server settings are correct. Use verbose/debug flags to see full SIP exchange.
2) Make a simple call (INVITE)
sipcli call sip:[email protected] --from sip:[email protected] --local-port 5060 This typically issues INVITE, handles provisional responses (100/180/183), awaits 200 OK with SDP, sends ACK, and sets up RTP if media is supported.
3) Hang up
To terminate:
sipcli hangup <call-id-or-dialog> # or during interactive call # Ctrl+C or sipcli bye 4) Send an instant message
sipcli message sip:[email protected] --from sip:[email protected] --body "Hello from SipCLI" 5) Send OPTIONS for capability discovery
sipcli options sip:sip.example.com --from sip:[email protected] Scripting and Automation
SipCLI shines when used in scripts. Example: automated outbound call test with logging and exit codes.
call_test.sh
#!/usr/bin/env bash TARGET="sip:[email protected]" USER="sip:[email protected]" PASS="testpass" LOG="/var/log/sipcli/call_test.log" sipcli call "$TARGET" --from "$USER" --password "$PASS" --timeout 20 --log "$LOG" STATUS=$? if [ $STATUS -eq 0 ]; then echo "Call succeeded" >> "$LOG" exit 0 else echo "Call failed with status $STATUS" >> "$LOG" exit 2 fi Use cron or monitoring systems to run periodic tests and alert on failures.
Media handling, codecs, and RTP
SipCLI can handle media in several ways depending on features compiled in:
- Generate/receive RTP streams using built-in tone generators for test calls.
- Relay RTP by interacting with system-level RTP engines (rtpengine, mediasoup) or connect to softphones.
- Support for common codecs: PCMU (G.711u), PCMA (G.711a), opus, G.722 — availability depends on build options and libraries.
Specify codecs and SDP overrides:
sipcli call sip:[email protected] --codecs PCMU,PCMA --sdp-file custom.sdp To test audio quality, pair SipCLI with sip-monitoring tools like sipp, rtcpstats, or call-quality metrics collectors.
Security: TLS, SRTP, and Authentication
- Use TLS for SIP over TLS (SIP-TLS) by providing CA certificates and enabling TLS mode:
sipcli register --server tls:sip.example.com:5061 --tls-ca /etc/ssl/certs/ca.pem - For media encryption, enable SRTP if SipCLI is built with libsrtp:
sipcli call ... --srtp aes_cm_128_hmac_sha1_80 - Support for digest authentication (most common) and, if available, more advanced auth like TLS client certs or OAuth-based access tokens.
Logging, Debugging, and Packet Capture
- Increase verbosity to see full SIP dialogs:
sipcli --verbose register ... sipcli --debug call ... - Output SIP message dumps to a file:
sipcli --dump sip-messages.log call ... - Use tcpdump/wireshark for RTP and SIP packet capture:
sudo tcpdump -i eth0 -w sip_capture.pcap port 5060 or port 5061Open the pcap in Wireshark, filter by sip or rtp, and inspect headers, SDP, and packet timing.
Common Issues and Solutions
- Authentication failures: verify realm, username, password; check for case-sensitivity and special characters needing shell escaping.
- NAT traversal: SIP over UDP often fails behind NAT. Use STUN/TURN/ICE or run SIP over TCP/TLS and configure symmetric RTP or an RTP relaying proxy.
- Codec mismatches: ensure both endpoints have a common codec; force codecs in SipCLI if needed.
- TLS handshake failures: check CA trust chain, certificate subject names, and supported TLS versions/ciphers.
Advanced Use Cases
- Load testing: use SipCLI in loops or parallel processes to generate SIP traffic for functional load tests. For large-scale load testing, combine with tools designed for high concurrency (sipp) while using SipCLI for targeted functional checks.
- Integration with CI/CD: run registration and basic call tests as part of deployment pipelines to validate SIP service health before cutover.
- SIP header manipulation: add or modify headers for testing server behavior:
sipcli call ... --add-header "X-Debug: sipcli-test"
Example: End-to-end registration and call script
demo_flow.sh
#!/usr/bin/env bash SERVER="sip.example.com" USER="alice" PASS="s3cr3t" TARGET="sip:[email protected]" sipcli register --server "$SERVER" --user "$USER" --password "$PASS" --realm example.com --verbose if [ $? -ne 0 ]; then echo "Registration failed" >&2 exit 1 fi sipcli call "$TARGET" --from "sip:$USER@$SERVER" --local-port 5062 --verbose --timeout 30 RET=$? if [ $RET -eq 0 ]; then echo "Call success" else echo "Call failed with code $RET" fi Resources and Further Reading
- SIP RFCs (RFC 3261 and related): read for protocol-level understanding.
- SDP RFCs for media negotiation.
- RTP/RTCP and SRTP specifications for media transport and security.
- Community forums and project-specific docs for SipCLI (flags, build options, platform-specific notes).
Conclusion SipCLI is a practical tool for anyone needing quick, scriptable SIP interactions without the overhead of GUI softphones. It’s ideal for testing, automation, and low-level debugging. Once you get comfortable with its commands, you can integrate SipCLI into monitoring, CI/CD, and diagnostic workflows to keep SIP services reliable.
If you want, I can: provide a ready-to-run script tailored to your SIP server credentials, explain NAT/STUN/TURN setup for SipCLI, or convert examples into Windows command or PowerShell equivalents.
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