On Tuesday 14 December 2004 15:19, Shoval Tomer wrote: > As far as I can remember I only opened sip and tftp ports for the phone. > > For some reason (didn't look into it too much) the call stays with sip > and doesn't use RTP. >
SIP is what sets up the session (ie it does session handling) RTP is the transport protocol that the audio uses. If you're using SIP then you're using RTP eos. Jon _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
