[Serusers] Bandwidth question

[Klaus Darilion]

SIP signaling goes through ser, RTP audio will be sent directly from SIP
UA to SIP UA (except you use rtpproxy for NAT traversal).

Klaus

> -----Original Message-----
> From: jerk face [mailto:jerkface2098 at yahoo.com] 
> Sent: Wednesday, December 17, 2003 4:57 PM
> To: serusers at lists.iptel.org
> Subject: [Serusers] Bandwidth question
> 
> 
> Hello.
> I was just curious how SER handles calls with its own
> members.
> For example: I want to have around 10 users on my SER.
>  If one calls the other, does the bandwidth get
> channeled through my server, or is the call handed
> off, directly connecting the users and saving my
> bandwidth?
> 
> 
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