06-22-2007 04:01 AM - edited 03-13-2019 04:15 PM
Hello,
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
06-28-2007 05:33 AM
07-19-2007 04:01 PM
Hi,
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.
Saludos..
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
fair-queue
!
int ser 0/1/0
service-policy out QoS-Policy
08-07-2007 10:55 AM
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.
Configure
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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