[Asterisk-Users] bandwidth
Eric Wieling aka ManxPower
eric at fnords.org
Mon Apr 4 19:39:14 MST 2005
Actually about 80k-82k when you take into account UDP and RTP overhead
and assume you are using SIP. Single IAX2 call may be a little less.
multiple IAX2 calls using trunking will be a lot less.
In fact, this question is answered on
http://www.digium.com/index.php?menu=documentation
specifically the link to
http://www.packetizer.com/voip/diagnostics/bandcalc.html
Unfortunatly the above URL is not terribly clear and understandable.
People complain about Asterisk's lack of good, organized, understandable
documentation. It might help if they actually used the documentation
and links that ARE available.
Here we have an example of one person that didn't do the research
(understandable, since he/she might not have known about the
Documentation link on Digium's web site) and then asked a question and
then another person that ALSO didn't do the research (I'm guilty of this
too, but am getting much better) but answered the question anyway.
William Boehlke wrote:
> The simple answer is 64KB.
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bernie
> Sent: Monday, April 04, 2005 4:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] bandwidth
>
> how much bandwith is used to go between a phone set and the asterisk server
> when a call is in progress? Just trying to plan out a system and need some
> figures to plan on bandwidth allocation.
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