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@yahoo.com]
Sent: Wednesday, December 17, 2003 4:57 PM
To: serusers(a)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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