On 06/19/10 15:19, Kamonwat Sookkara wrote: > > Dear Asterisk friends, > > > > Please help me to clarify my doubt. After monitor SIP and RTP > traffic with Wireshark. I found that both SIP and RTP traffic between > 2 sip clients must be passed through Asterisk. > > Is it possible that 2 sip clients connect with each other directly > for RTP session after sip session completed ? >
By default it is yes, however within a LAN environment you can usually allow clients to re-invite directly between themselves. Check the "canreinvite" option out.// -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
