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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.21s
  • Memory usage: 2.28MB
  • Database queries: 29
  • GZIP: Disabled
  • Server load: 2.21

How To Debug and Troubleshoot VOIP

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
  • many other tools available for this function, add your favorite to this list!
  • 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.






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

  • 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



Monitoring and Test Tools





Asterisk Tools

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

Network Impairment Simulators

  • Spirent Communications - Test Solutions for VoIP networks and devices
  • 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.
  • Apposite Technologies Linktropy 4500 hardware appliance to emulate WAN bandwidth, delay, and loss up to 155 Mbps.
  • Candela Technologies LANForge ICE Network Emulator
  • UDP Packet Reflector and Forwarder open source tool that can drop packets, duplicate packets, and add jitter on a per port basis.
  • 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
  • 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!



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



Protocol Debug


Other Sites


  • CallManager Express Resource - design, tips and tricks and troubleshooting help for Cisco's SMB voice platform.


See also


Created by jht2, Last modification by mark_stacy on Fri 16 of Mar, 2007 [13:56 UTC]

Comments

Comments Filter

Quintum install

by Ahmad on 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

K-312GWT

by fernando on 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

www.voiptroubleshooter.com

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

NIST Net

by Anonymous on 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.

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