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@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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Klaus Darilion