Well, this should give you enough to chew on since voice is becoming a hot topic. I am trying to configure VoIP with QoS. Why over IP and not over ATM, you say? I have to controll the call with a H.323 Gatekeeper, and that is IP.
My problem appears to be that the call setup (or maybe signalling?) appears to be delayed. The test results are as follows: If the WAN link is saturated with data packets PRIOR to establishing the voice call, the first 10 to 15 (approximately) seconds of the call are lost. After the call is established, voice is rock solid and no voice packets are delayed or lost. If the voice call is established PRIOR to saturating the WAN link with data packets, the voice call is rock solid and no voice packets are delayed or lost. Thoughts or configs would be appreciated. --Mark version 12.2 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname Router ! logging buffered 4096 debugging ! memory-size iomem 25 ip subnet-zero ! no ip domain lookup ! ip cef ! voice call carrier capacity active voice rtp send-recv ! no voice hpi capture buffer no voice hpi capture destination ! vc-class atm vip vbr-rt 256 256 10 precedence 5 no bump traffic no protect vc no protect group ! vc-class atm normal vbr-nrt 192 192 precedence other no protect vc no protect group ! interface ATM0/0 ip address 1.1.1.254 255.255.255.0 ip nat outside no atm ilmi-keepalive bundle-enable bundle qosmap protocol ip 1.1.1.1 encapsulation aal5snap pvc-bundle data 0/37 class-vc normal pvc-bundle voice 0/36 class-vc vip ! dsl equipment-type CPE dsl operating-mode GSHDSL symmetric annex A dsl linerate AUTO h323-gateway voip interface h323-gateway voip id Gatekeeper ipaddr x.x.x.x 1718 h323-gateway voip h323-id Gateway ip rsvp bandwidth 64 64 ip rsvp resource-provider wfq pvc ! interface FastEthernet0/0 ip address 10.200.100.1 255.255.255.0 ip nat inside speed auto ! ip nat inside source list 1 interface ATM0/0 overload ip classless ip route 0.0.0.0 0.0.0.0 1.1.1.1 no ip http server ip pim bidir-enable ! access-list 1 permit 10.200.100.0 0.0.0.255 ! call rsvp-sync ! voice-port 2/0 station-id name StaID station-id number 1112223333 caller-id enable ! voice-port 2/1 station-id name StaID station-id number 1112223333 caller-id enable ! dial-peer cor custom ! dial-peer voice 1 voip destination-pattern T session target ras ! gateway ! line con 0 line aux 0 line vty 0 4 login ! no scheduler allocate end Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=57104&t=57104 -------------------------------------------------- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
