> 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
