In this day of companies downsizing and more employees working from home to cut office space costs, voice and video traffic over the internet is becoming more prevalent and also critical that it the quality is reliable.
We have several employees who work out of their homes and in an effort to maintain acceptable voice quality, we have configured QoS on their routers.
RTP and RTCP traffic is prioritized. We created an access group called PlixerVOIP which bundles SIP (5060 -Session Initiation Protocol) and RTP (range 10000 20000), then set the DSCP value to EF, giving it 70% of the bandwidth.
ip access-list extended PlixerVoIP permit udp any any eq 4569 permit udp any any eq 5060 permit udp any any range 10000 20000 ! policy-map TP-Pol-FastEthernet0/1 class PlixerVoIP set dscp ef priority percent 70 class Web_Email bandwidth remaining percent 75 class class-default fair-queue interface FastEthernet0/1 ip dhcp client client-id ascii dalel ip dhcp client class-id ascii ip dhcp client hostname dalel.plixer,com ip address dhcp no ip proxy-arp ip nbar protocol-discovery ip flow monitor nbar-mon input ip flow monitor nbar-mon output ip flow ingress ip nat outside ip virtual-reassembly in duplex auto speed auto service-policy output TP-Pol-FastEthernet0/1 service-policy type performance-monitor input performancemonitor service-policy type performance-monitor output performancemonitor
We can then monitor voice quality with Cisco’s Medianet and Scrutinizer NetFlow traffic monitoring.
Notice in the Cisco Medianet Jitter report above how you can easily see jitter and packet loss information, real time!
Prioritizing VOIP traffic on the remote user’s router minimizes the choppiness of the voice traffic, providing better service to our customers as an end result.
Just one more feature provided by our Best at NetFlow solution!
