Help to setup Qos for voip connection

Unanswered Question
Jun 22nd, 2007


We are in the process of implementing a ToIP with CME (over SIP) for our WAN links so that we can run VoIP between 14 locations and I have lots of questions regarding how to configure QoS.

We have a link between location with LL 2Mega for all locations.

In each location , we will be using one IP Phone and media gateways for voice communications and data (for data we have a critical applications).

i'm planing to use the G729 codecs with 2 calls simultaneous between locations and central site.

How would I begin to configure QoS for these network connections??

thank you in advance

I have this problem too.
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jaregalado Thu, 07/19/2007 - 16:01


We have deployed multiple VoIP sites with QoS working fine for us using

this QoS configuration on our routers, you may need to modify it to

suit your specific needs.


ip access-list extended VoIP-Control

remark H.323 Slow Start

permit tcp any any range 11000 11999

remark H.323 Fast Start

permit tcp any eq 1720 any

remark IP Phone Skinny Protocol

permit tcp any any eq 2000

remark IP Phone SIP

permit tcp any any eq 5060

permit udp any any eq 5060

ip access-list extended VoIP-RTP

remark Canales de voz por RTP/UDP

permit udp any any range 16384 32767

class-map match-any VoIP-Control

match access-group name VoIP-Control

class-map match-any VoIP-RTP

match protocol rtp audio

match access-group name VoIP-RTP



policy-map QoS-Policy

class VoIP-RTP

priority percent 70

set dscp ef

class VoIP-Control

set dscp af31

bandwidth percent 5

class class-default



int ser 0/1/0

service-policy out QoS-Policy

dford333 Tue, 08/07/2007 - 10:55

I added some additional info since the previous response gives you the required QOS settings

you not only providing QOS but also call Admission Control (CAC)

Also make sure that location are set right on your remote site for 2 call

and unlimited on you central site.

Refer to Unified CallManager Interactive Voice Network Configuration and Troubleshooting Case Study for requirements assumed by this series of interactive documents.


In order to configure CAC, complete these steps:

1. Configure Locations and Bandwidth

2. Configure an Alternative Automatic Rerouting (AAR) Group

3. Enable AAR in Unified CallManager

4. Configure AAR Calling Search Space on the IP Phones

5. Ensure that IP Phones Contain an External Phone Number Mask

Note: AAR uses the prefix configured under the AAR group and the external phone number mask of the destination phone in order to construct the full E164 number.

For example, if phone 1 (5005) calls remote phone 2 (4002), phone 1 uses prefix 91, as configured under the AAR group, and adds it to the mask for the remote phone, which is 9195014002. The number becomes 919195014002.

Make sure that you do not include the long distance digit (1) in the external phone number mask; otherwise, the number is included as part of the caller ID.

Configure Locations and Bandwidth

Configure locations and bandwith per remote location as described in this section.

1. Click System > Location > Add a New Location.

2. Configure the location as shown in this image:

This example uses the default location of None for the San Jose head office. The bandwidth for the head office is left at default and unlimited. For the remote location (RTP), this example configures a location Remote.

Note: This example configures a bandwidth of 48 kbps, which allows 2 x G.729 call over the WAN. This does not include Layer 2 overhead. For information on how different codec use different bandwidth values, refer to the Call Admission Control chapter of the Cisco Unified Communications SRND Based on Cisco Unified CallManager 4.x.

Configure an Alternative Automatic Rerouting (AAR) Group


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