Looks fine to me, you only need to specify DSCP or TOS (may need to check the manual for which it takes first).
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Johnson Sent: Wednesday, December 05, 2007 14:02 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisk server and DSCP QOS Thanks, Darryl, To clarify: in /etc/asterisk/sip.conf you have the lines: tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. and in your Polycom configuration [I'm using Polycom's sip 2.2.0] you have something like (this is the one I'm uncertain about): <QOS> <Ethernet> <RTP qos.ethernet.rtp.user_priority="5"/> <CallControl qos.ethernet.callControl.user_priority="5"/> <Other qos.ethernet.other.user_priority="2"/> </Ethernet> <IP> <RTP qos.ip.rtp.dscp="184" qos.ip.rtp.min_delay="1" qos.ip.rtp.max_throughput="1" qos.ip.rtp.max_reliability="0" qos.ip.rtp.min_cost="0" qos.ip.rtp.precedence="5"/> <CallControl qos.ip.callControl.dscp="184" qos.ip.callControl.min_delay="1" qos.ip.callControl.max_throughput="0" qos.ip.callControl.max_reliability="0" qos.ip.callControl.min_cost="0" qos.ip.callControl.precedence="5"/> </IP> </QOS> Thanks again! Steve Darryl Duncan wrote: We're using 184 here (aka TOS 5/EF). Not set by iptables though, instead it is set in sip.conf (tos_sip/tos_audio) and on our Polycom/Cisco phones. -----Original Message----- Subject: [asterisk-users] Asterisk server and DSCP QOS Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using the Linksys SGE2000P POE switch which supports QOS via DSCP. Thanks a lot for any info. Steve _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users