Mark,
It sounds like the voice traffic is being prioritized correctly since
the voice is "rock solid" after the connection is made. Looking at your
config this is strange as the default IP Precedence should be 0 for
voice traffic and I do not see where you have specified this in your
dial peer. Based on your explanation I would expect to see something like:
dial-peer voice 1 voip
destination-pattern .....
session target ...
ip qos dscp cs5 media
-or-
dial-peer voice 1 voip
destination-pattern .....
session target ...
ip precedence 5
With the classification statements missing from your configuration, I
would expect voice to be choppy and unintelligible. The default
classification for router-originated voice traffic is supposed to be ip
precedence of 0 (DSCP=000000). Perhaps Cisco has changed this in the
latest IOS releases?
You should also have "ip qos dscp cs5 signaling" present in your
dial-peer configuration to identify the signalling traffic as having IP
precedence=5 so that it is classified correctly and sent down your voice
PVC. As it stands now, your signalling traffic should be using your
data PVC.
You can find some decent QoS configuration examples on Cisco's web site
relating to LLQ which might also help along these lines. Cisco usually
recommends setting signaling to af31 and media to ef (to make sure that
if signaling and media contend for bandwidth voice quality is not
affected). Since your configuration is already classifying explicitly
on ip precedence=5, it might just be simpler to set the ip precedence of
both media and signaling traffic to 5.
I hope this helps. :)
- Tom
Mark S wrote:
> For those of you trying to email me from the link in the message, here is
> the updated post. Sorry about the duplicate.
>
> *******************
> 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
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