[Asterisk-Users] bandwith calculation

[Dan Fernandez]

Messagethanks Frank.

Calculating the bandwith with these tools shouldn´t be hard, I assume. The other part of my question is what has me more worried. That is, are  there any instances where depending on the codec (or some other factors) the RTP stream also goes through * rather than just from A to B. Any ideas?

  ----- Original Message ----- 
  From: Frank Hoonhout 
  To: asterisk-users at lists.digium.com 
  Sent: Friday, April 25, 2003 12:31 PM
  Subject: RE: [Asterisk-Users] bandwith calculation


  I found these on these.

  Web based
  http://www.packetizer.com/iptel/bandcalc.html

  Window's executable program (my favorite)
  http://www.networkingfiles.com/Network/clarent.htm

  Frank...
    -----Original Message-----
    From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Dan Fernandez
    Sent: Thursday, April 24, 2003 1:28 PM
    To: asterisk-users at lists.digium.com
    Subject: [Asterisk-Users] bandwith calculation


    I would like to know how to calculate the amount of bandwith I would need to host X number of calls.

    For example, if user A in San Francisco with an ATA 186 calls user B in New York with an ATA 186 and Asterisk is being hosted in a PC in Miami. How much bandwith do I need to have in Miami?  Do I just need bandwith for the setup of the call (ie the SIP part) or are there any instances where the RTP stream goes through Miami as well?  



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