So the more reliable way to do QoS is with MAC adress and not on a port basis. Am I right ?
Thanks for your help, Adrien On 9/19/05, Rich Adamson <[EMAIL PROTECTED]> wrote: > > > I know that SIP is using port 5060 for session initiation, but which port > > does it use for audio ? is it dynamically assigned ? > > Its dynamically assigned on a per-call basis. > > Asterisk assigns the port based on contents of rtp.conf. > > Remote sip phones assign port numbers based on whatever the manufacturer > happened to choose (no industry standard). E.g., Cisco uses 32,768 to > something around 40,000, while xlite uses something in the area of 8,000. > The various manufacturers are not consistent at all. > > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 > -- Adrien Laurent [EMAIL PROTECTED] www.modulis.ca _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- 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
