[Asterisk-Users] sip/rtp performance monitoring

[James H. Thompson]

If the customers are using an ATA or other VOIP device that supports RTCP, then you can often get packet loss and jitter stats by
extracting the RTCP packets and analyzing them.
This will actually give you the packet loss and jitter that the customer is seeing in the received RTP stream from you.
A combination of Tetheral and grep or perl can get you along way in capturing and analyzing this data.


Jim

James H. Thompson
jht at lj.net

  ----- Original Message ----- 
  From: Forrest Christian 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Saturday, August 06, 2005 9:43 AM
  Subject: [Asterisk-Users] sip/rtp performance monitoring


  I'm currently running asterisk to provide VoIP services to clients of 
  the ISP I work for.

  I would like to be able to tell if I am loosing packets and/or are 
  having other issues with any of the voice streams, so I can address them 
  proactively.

  I'm not particularly interested in spending oodles of money buying one 
  of the commercial analysis tools.   Is there some open source tool (or 
  something I can monitor in asterisk) which will tell me if I'm missing 
  packets or similar?  I realize this will likely be only from the 
  customer towards me since I can't really monitor at the customer end.

  -forrest
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