How To Debug and Troubleshoot VOIP

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Debugging and troubleshooting VOIP problems.

(SIP, MGCP, H.323, RTP, Skinny etc.)

One of the primary techniques is to view what is actually getting sent and received by VOIP devices. There are several ways to do this:
  • Monitor Ethernet Traffic
  • Debugging displays from a VOIP program

It helps to understand whats supposed to be happening. Studying the relevant RFCs and other protocol documents and tutorials is helpful.

Ethernet Monitoring Tools

  • ClarifiedNetworks
    • HowNetWorks - a free VMWare appliance
    • Tia - sophisticated monitoring and flow analysis tool with visualisation, multiple data sources, ...
  • List of Monitoring and sniffing software
  • ngrep (Available for Linux, Windows, Apple, BSD, etc.)
    • Dumps only the ASCII portion of packets, excellent for ASCII based protocols
  • sngrep (Available for Linux, Apple, BSD.) by Irontec
    • Curses based SIP dialogs monitoring tool.
    • Support UDP, TCP and TLS.
    • Graphical(curses) dialog summary and dialog flow detail.
    • Very useful for debugging and learning purposes.
    • GPL License
  • Packetyzer: User-friendly packet sniffer for Windows, supports SIP
  • Rate which provides real time packet-per-second and data transfer rates
  • SIP Workbench Displays SIP ladder diagram from WireShark/pcap captures
    • Displays STUN/TURN interactions
    • Allow users to filter on particular call flows
  • Spirent Communications - Test Solutions for VoIP networks and devices
  • STINGA SS7 Protocol Analyzer and Monitoring System
    • Call trace, mature SS7 protocol decodes
    • SIP, SS7, SIP-T, SIGTRAN, ISDN, MGCP and Megaco/H.248
    • Performance analysis
    • Open MySQL database
  • TamoSoft CommView - network analyzer for Windows
    • Real-time VoIP call monitoring
    • SIP and H.323 analysis and decoding, call playback
    • Jitter, QoS, Bandwidth charts
  • tcpdump (standard utility in most Linux distributions)
  • Touchstone
    • WinEyeQ
      • 100% software-based
      • monitors/analyzes/records/replays SIP and H.323 traffic, audio/video media and QOS.
    • TraceBuster (free version available)
      • records/replays SIP and H.323 traffic, audio/video media and QOS.
  • VoIPmonitor is open source live network packet sniffer which analyze SIP and RTP protocol.
    • Predicts MOS-LQE score according to ITU-T G.107 E-model
    • Detailed delay/loss/MOS statistics stored to MySQL
    • Each call is saved as standalone pcap file
    • Possibilities to convert pcap to WAV
  • WinDump - tcpdump for Windows
  • Wireshark (Ex Ethereal) (Open Source and available for Linux, Windows, Apple, BSD, etc.)
    • Support for decoding many VOIP protocols is included (including IAX)
    • VoIP call analysis and call flow diagram for (SIP, MGCP, H323, etc.)
    • RTP Statistics and graph analysis (jitter, delay, packet lost, etc.)
    • RTP playback (Wireshark only)
  • SIPDump - an open source c# tool for logging calls to pcap files. SIP and RTP. Very early development. Runs on windows or Linux with mono.
  • StarTrinity SIP Tester - Freeware SIP and RTP monitoring tool with G.107 MOS/R-factor, RFC3550 global max jitter, realtime charts and reports.

Built-in Debug Tools

  • Asterisk Use the sip debug command
  • linphone Outputs useful diagnostics to the console as it uses the oSIP library
  • pjsua A command line SIP user agent from pjsip.org, available for various platforms, and very useful to debug SIP functionalities (call, presence, instant messaging, etc.) as well as media quality via RTCP statistics.
  • xten The x-lite and x-pro SIP soft phones have a buit-in display and decoding of received and sent SIP packets (Hit F9 to activate)
  • miniSipServer has inner trace tools to capture all SIP messages.

Traffic generators

  • Candela Technologies LANForge FIRE VOIP/RTP/PESQ call generator
  • Codima autoVoIP Blaster - Ensure VoIP Readiness - Stress Test Network Limits
  • Empirix Signaling and Media load and feature testing
  • GL Communications
    • PacketGen - generates SIP calls with or without RTP traffic
    • PacketScan - monitor, collect, and analyze QoS statistics on VOIP traffic
  • Integrated Research Prognosis will simulate, record and analyze VOIP traffic in real time.
  • Iperf creates network traffic and measures performance
    • Can be used to test a network to see how it might perform with increased VOIP traffic
  • Ixia VOIP traffic generators and Network assessment tools
  • SIP Inspector Pro combines the best when in it comes to functionality and ease of use. This tool is a must have when it comes to working with SIP.
    • Can create custom SIP signaling scenarios, messages
    • Can play RTP from saved pcap file
    • Unconditionally streams RTP
    • Can act as a client and a server
  • Nexus Telecom Load testing, modular system
    • 380,000 calls per hour, per module
    • 4Mbps RTP real media per module
    • Up to 400,000 simulated subscribers
  • MyVoIPSpeed simulates VoIP traffic over your Internet connection, measures key diagnostics including Jitter and Packet Loss, and provides an analysis of the voice quality
  • PacketIsland 4"x4" in-line micro-appliances used in a distributed multi-site enterprise or SME to generate live VoIP traffic and measure loss, jitter, MOS, route performance, route flaps, etc. Also measures ongoing data traffic in network.
  • pjsip-perf Open source call generator from pjsip.org to measure SIP call/transaction performance.
  • Sipp SIP Performance Test Tool - Performance tester for SIP
  • Spirent Communications - Test Solutions for VoIP networks and devices
  • Touchstone 100% software-based VoIP and video verification tools.
    • WinSIP - SIP signaling and Audio/Video media generator
    • Win323 - H.323 signaling and Audio/Video media generator
  • Valid8.com Valid8.com is a leading provider of SIP, H.323, Megaco, SIGTRAN traffic generation solutions.
  • ivrworx - high level Lua interface to SIP/RTSP/MRCP, for testing distributed VoIP scenarios (windows, Vista+ clients).
  • StarTrinity SIP Tester - bulk SIP call generator with complex test scenarios specified as a script, up to 5400 simultaneous G.729 calls per single server. G.107 MOS, PESQ MOS measurement, email alerts and daily reports.
  • StarTrinity VoIP Network Tester - freeware, generates traffic with multiple UDP streams and measures VoIP network jitter and delay.



Monitoring and Test Tools


See Also:


Asterisk Tools

  • Nocom A simple script for viewing Asterisk config files with comments and empty lines removed.

Network Impairment Simulators

  • Apposite Technologies Linktropy 4500 hardware appliance to emulate WAN bandwidth, delay, and loss up to 155 Mbps.
  • Candela Technologies LANForge ICE Network Emulator
  • IPWave simulates many types of network impairments
  • iTrinegy offers a full range of network application performance testing tools
  • NIST Net allows a single Linux PC set up as a router to emulate a wide variety of network conditions
  • Shunra Network Simulator Shunra Virtual Enterprise (Shunra VE) network simulator creates a model of any production environment, including the network, remote locations, and the number and distribution of local and remote end-users. With Shunra VE, users can test the functionality, performance, scalability and robustness of the VoIP infrastructure under current and future production conditions — before deployment over the network.
  • Simena Network Emulator hardware appliance can simulate just about any possible network condition including latency, bandwidth, congestion, packet loss, etc. Test where your VoIP will break!
  • Spirent Communications - Test Solutions for VoIP networks and devices
  • UDP Packet Reflector and Forwarder open source tool that can drop packets, duplicate packets, and add jitter on a per port basis.

Decoding VOIP audio streams

There are several approaches to converting an RTP stream of packets into a playable audio.
See: Converting RTP to audio

SIP Debug

  • Callflow - creates a diagram of SIP flows
  • Distributed SIPFlow - Distributed application for capturing and displaying SIP callflows.
  • Free VoIP TraceBuster Decode and analyze Ehtereal/WireShark capture files!
  • PacketIsland 4"x4" in-line micro-appliances can capture SIP traffic and display ladder graphs of SIP call flow. SIP data from thousands of calls can be stored for weeks or months.
  • SIPFlow Standard - Java tool for displaying SIP traffic captured in real-time, or imported from Ethereal or tcpdump
  • SIP Grapher ( sip_grapher.pl ) Builds SIP flow graphs from Asterisk log files with sip debug enabled - search by phone number or Call-ID. Written in Perl.
  • Sipsak Command line utility for testing SIP devices and Programs
  • SIP Scenario - creates a diagram of SIP flow
  • siptest a command line test tool for sending and receiving SIP messages
  • sipviewer a visual SIP message trace tool
  • Wireshark (Ex Ethereal) using the VoIP call analysis feature.

Cisco Troubleshooting Tools

  • SMART-Smart Message Architecture Reader & Translator The most advanced audio and video protocol log parser in the world, capable of parsing and analyzing output logs from enterprise platforms such as Cisco Unified Communications Manager, Cisco Telepresense Systems, Cisco Telepresence Video Communication Server, Cisco IOS, and others. SIP Troubleshooting, H323 Troubleshooting and Q931 Troubleshooting.

Protocol Debug


Other Sites


See also



Created by: jht2, Last modification: Mon 24 of Nov, 2014 (07:40 UTC) by irontecvoip


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