VOIP and QOS

Hi, have limited knowledge of QOS and need to configure QOS on our

2950s that's running our phone system asterisk, we are using a combination of polyocm and aastra sip phones. I hope like to configure QOS on and get rid of our current choppy calls - anyone can give me an idea or example of the commands I will use for QOS?
Reply to
mikeg001
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add this to your phone int. works fine for us. Also you will need qos on router as well.

switchport priority extend cos 0

Reply to
luvsmecats

Do i apply that to all my ports on my switch ?

Reply to
mikeg001

I think based on what i have seen this command is for cisco ip phones or will that work on any ip phone ?

Reply to
mikeg001

Reply to
luvsmecats

Originally, I enabled AutoQoS.

Then, seeing that the automatic QoS configuration had mls qos trust cos, and all DSCP codes were being lost through trunk ports, I changed each interface to this: mls qos trust dscp

After that, I was asked to configure QoS to specifically identify voice traffic and give it a high DSCP code and a 0 DSCP code for all other traffic, so I did this:

! class-map match-all voice match access-group name voice.traffic ! policy-map Assign.DSCP class voice set dscp ef ! ip access-list extended voice.traffic permit ip 10.20.0.0 0.0.255.255 any ! interface FastEthernet1/0/14 switchport trunk encapsulation dot1q switchport mode trunk service-policy input Assign.DSCP srr-queue bandwidth share 10 10 60 20 srr-queue bandwidth shape 10 0 0 0 auto qos voip trust spanning-tree portfast !

Having said that, if you don't know *why* the voice is "choppy", then chances are that QoS *won't* fix it - 99% of voice-quality problems will be caused by bad configuration of the voice system itself, and not the network (until you come to put it across WAN links).

Get Wireshark onto the network and capture an entire phone call - the summarised rtp stream details will show you the number of lost packets (if any) as well as the average & maximum delays and jitter.

Check your jitter-buffer settings are comfortably higher than the jitter shown in the rtp stream summary.

Reply to
Arthur Brain

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