05-10-2006 11:48 PM - edited 03-17-2019 08:41 PM
I have a CME 3.2 router talking to a CM 4.0 server via a H323 dial-peer. They are connected via a 512/256K. I have the following QOS statement, and config.
class-map match-all VOICE-CONTROL
match ip dscp af31
match dscp cs3
class-map match-any VOICE
match ip dscp ef
policy-map VOIP-QOS
class VOICE
priority 96
class VOICE-CONTROL
priority 8
class class-default
fair-queue
interface FastEthernet0/0
bandwidth 256
ip address 172.18.33.2 255.255.255.0
service-policy output VOIP-QOS
interface FastEthernet0/1.1
encapsulation dot1Q 1 native
ip address 10.181.8.253 255.255.255.0
!
interface FastEthernet0/1.2
encapsulation dot1Q 2
ip address 10.181.9.253 255.255.255.0
interface FastEthernet0/1.3
encapsulation dot1Q 3
ip address 10.181.10.253 255.255.255.0
h323-gateway voip bind srcaddr 10.181.10.253
ip route 10.140.0.0 255.255.255.0 172.18.33.1
ip route 10.140.1.0 255.255.255.0 172.18.33.1
ip route 10.140.12.0 255.255.255.0 172.18.33.1
ip route 10.140.14.0 255.255.255.0 172.18.33.1
ip route 10.140.99.0 255.255.255.0 172.18.33.1
ip route 10.141.0.0 255.255.0.0 172.18.33.1
ip route 10.150.1.0 255.255.255.0 172.18.33.1
ip route 10.253.2.0 255.255.255.0 172.18.33.1
dial-peer voice 4000 voip
destination-pattern 4[136-8]..
session target ipv4:10.141.1.1
dtmf-relay h245-alphanumeric
no vad
telephony-service
load 7910 P00403020214
load 7960-7940 P00305000600
max-ephones 30
max-dn 100
ip source-address 10.181.10.253 port 2000
timeouts interdigit 3
create cnf-files version-stamp Jan 01 2002 00:00:00
max-conferences 4
transfer-system full-consult
ephone-dn 1 dual-line
number 3799
ephone 1
mac-address 0017.5987.88F7
type 7960
button 1:1
The phones are all on Vlan 3, with the PC's and servers on Vlan 2. With no traffic on the link, Voice Quality is fine, but when outbound traffic is maxing the link, the qos statements appears to be working, if I do a sh policy-map interface fastethernet 0/0. But I am still getting a lot of jitter on the call, for the oubound voice from the CME box. eg, Users at the CM end hear the jitter, while users at the CME do not.
I have tired everything I can find to do with QOS and Voice traffic, and at a loss. Does anybody have any fresh ideas that I can try, or see anything obvious that I have wrong.
05-11-2006 06:36 AM
Hello,
there are a few recommendations and fresh ideas.
First, your queueing (f.e. "priority 8") will not have any effect in the config shown. The underlying reason is, that LLQ is only involved, when the hardware queue is overloaded. In your case this would mean that there is more traffic than your FE0/1 can handle!
So you need a nested policy to throttle the traffic to the desired value of 512k. Or you implement your policy at the router with your WAN link. Assuming this might not be possible you could do the following:
class-map match-all VOICE-CONTROL
match ip dscp af31
match dscp cs3
class-map match-any VOICE
match ip dscp ef
policy-map VOIP-QOS
class VOICE
priority 96
class VOICE-CONTROL
bandwidth 8
class class-default
fair-queue
policy-map shape256k
class class-default
shape 256000
service-policy output VOIP-QOS
interface FastEthernet0/0
bandwidth 256
ip address 172.18.33.2 255.255.255.0
service-policy output shape256k
This will make sure that there is no more traffic than the WAN link can handle and your service-policy is applied, when more than 256k of traffic crosses the FE0/1.
Second, jitter will be caused by large IP packets (1500 Byte) blocking your VoIP packets. Once the WAN router starts transmitting (OSI physical layer) a 1500 Byte IP packet (f.e. FTP download) it can not stop transmitting. Thus a VoIP packet might have to wait up to 47 ms (1500 Byte over 256k) - this gives a Jitter of 47 ms. The second contribution to jitter, presumably the larger one right now, comes from the queueing system on the WAN router, which does not prioritize VoIP packets (assumed). Thus even hundreds of ms could be observed.
Conclusion: modify the config to the suggested example config above. This should improve the situation. It might still give you jitter up to 50 ms because of large IP packet on a low bandwidth link (256k). Usually telephones can cope with this without massive voice quality degradation.
Hope this helps! please rate all posts.
Regards, Martin
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