On your sip.conf for each sip endopoint set canreinvite = yes. That way the rtp stream won´t go through *. The only problem though is for ATA 186. They need canreinvite = No when they are in a NAT environment.
----- Original Message ----- From: "Low, Adam" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 28, 2003 11:29 AM Subject: [Asterisk-Users] RTP session traversing Asterisk server ... > > I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would expect the RTP session should ideally be between the two end points of the call, in my case the AS5300 and the 7940 which are connected on the same VLAN as the Asterisk server. > > When I sniff the packets on the VLAN I find that all RTP packets are being relayed by the Asterisk server causing increased load on the server and ultimately a higher latency between the two end points. > > Is this a typical operation of Asterisk or is this possibly due to the fact that some of the phones (not those used in the tests) are running NAT and Asterisk relays all RTP packets ? > > Adam > > > ********* DISCLAIMER ********* > > This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users