On Wed, 2008-01-23 at 15:53 -0800, David Newman wrote: > How you detect a VoIP flow may also be an issue. If your VoIP traffic > uses SIP, you can classify the signaling traffic on 5060/udp -- but then > the voice or video traffic will use RTP/RTCP and some ephemeral port > chosen during call setup.
... > (If anyone has a method for RTP/RTCP awareness in pf -- including the > ability to set up and tear down rules for the call duration -- please > share it!) I am just wondering if the RTP proxy in siproxd could help. I guess one could write pf (altq) rules based on the RTP port range chosen. May not be so flexible or even suitable in every scenario (since one needs to setup a siproxd), then again... (See http://siproxd.sourceforge.net/index.php?op=faq for RTP proxy details.) What do you think?
