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How To Debug and Troubleshoot VOIP

Created by: jht2,Last modification on Fri 12 of Mar, 2010 [03:41 UTC] by jeffdoubleyou

Debugging and troubleshooting VOIP problems.

(SIP, MGCP, H.323, 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

  • 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)
  • Spirent Communications - Test Solutions for VoIP networks and devices
  • tcpdump (standard utility in most Linux distributions)
  • WinDump - tcpdump for Windows
  • ngrep (Available for Linux, Windows, Apple, BSD, etc.)
    • Dumps only the ASCII portion of packets, excellent for ASCII based protocols
  • Packetyzer: User-friendly packet sniffer for Windows, supports SIP
  • List of Monitoring and sniffing software
  • Rate which provides real time packet-per-second and data transfer rates
  • ClarifiedNetworks
    • HowNetWorks - a free VMWare appliance
    • Tia - sophisticated monitoring and flow analysis tool with visualisation, multiple data sources, ...
  • TamoSoft CommView - network analyzer for Windows
    • Real-time VoIP call monitoring
    • SIP and H.323 analysis and decoding, call playback
    • Jitter, QoS, Bandwidth charts
  • 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.
  • SIP Workbench Displays SIP ladder diagram from WireShark/pcap captures
    • Displays STUN/TURN interactions
    • Allow users to filter on particular call flows

Built-in Debug Tools

  • 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)
  • 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.

Traffic generators

  • Codima autoVoIP Blaster - Ensure VoIP Readiness - Stress Test Network Limits
  • Spirent Communications - Test Solutions for VoIP networks and devices
  • Candela Technologies LANForge FIRE VOIP/RTP/PESQ call generator
  • 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
  • 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.
  • Sipp SIP Performance Test Tool - Performance tester for SIP
  • pjsip-perf Open source call generator from pjsip.org to measure SIP call/transaction performance.
  • 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.


Monitoring and Test Tools



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
  • 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
  • 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.
  • 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.
  • 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
  • SIPFlow Standard - Java tool for displaying SIP traffic captured in real-time, or imported from Ethereal or tcpdump
  • Distributed SIPFlow - Distributed application for capturing and displaying SIP callflows.
  • Wireshark (Ex Ethereal) using the VoIP call analysis feature.
  • Free VoIP TraceBuster Decode and analyze Ehtereal/WireShark capture files!


Protocol Debug


Other Sites



See also



Comments

Comments Filter
222

333beep sound in voip audiocodes mp124

by g_raju_21, Friday 30 of May, 2008 [04:53:59 UTC]
hai! i am facing beep sound problem while conversation with other on voip audiocodes mp124 can any one help how to comeout this problem?
222

333InterTel Woes

by Shorehell, Thursday 06 of March, 2008 [18:37:26 UTC]
We have recently implemented a new firewall onsite. And now our remote IP phones cannot talk to the switch. We currently have the intertel/axcess 5000. I'm sure its just a port I need to open. Any Ideas?
222

333InterTel Woes

by Shorehell, Thursday 06 of March, 2008 [18:25:54 UTC]
We have recently implemented a new firewall onsite. And now our remote IP phones cannot talk to the switch. We currently have the intertel/axcess 5000. I'm sure its just a port I need to open. Any Ideas?
222

333Quintum install

by ahmad79, Tuesday 06 of February, 2007 [00:30:35 UTC]
Hello

I have Quintum AX i want somebody help me to configure and install it , i want work with me per houre remotely .

if anyone want work with me please contact me at MSN a3_3a@hotmail.com or send email for it .

Thanks

222

333K-312GWT

by fenarro, Monday 15 of May, 2006 [00:29:06 UTC]
hELLO, I AM TRYING TO CONFIG THE SIP K-312GWT , IT IS CONNECTED TO MY linksys router. Everything is conected ok, but i cant connect to the server. I use voip stunt, i think i must open a port in the router, wich one ???

Pleasse help
Fernando
222

333www.voiptroubleshooter.com

by willkemp, Saturday 30 of April, 2005 [06:55:46 UTC]
http://www.voiptroubleshooter.com/ now appears to be working fine with Opera and Firefox.
222

333NIST Net

by , Tuesday 18 of January, 2005 [11:00:59 UTC]
Network Impairment Simulators could be supplemented with NIST Net
http://www-x.antd.nist.gov/nistnet/
We use it regularly, and even though it's a bit difficult to use it serves its purpose very well.