Quality of Service, or QoS refers to the identification and prioritization of specific traffic on a network. How QoS is implemented on a network is very dependent on the specific network hardware and software on that network. Below are some general guidelines that will assist you in setting up your network for High Quality Voice.
Bandwidth Considerations for Inbound and Outbound Traffic
In today's modern networks, voice services have relatively low bandwidth requirements. A single voice call using the most common codec will take around 100kbps per concurrent call. If you need to support 10 concurrent calls, then no less then 1000kbps, or 1mbps will be required for voice services.
Care should however be taken to not under-size an ISP connection that is going to support voice services. While voice services take up a relatively small amount of bandwidth, other network activities such as file downloading and video streaming can quickly consume all of the available download bandwidth to your site. If this happens, excess traffic above your allowed download bandwidth will be discarded by your ISP. Any calls in progress while traffic is being discarded will have voice quality issues.
Due to regulatory issues, ISPs are not allowed to modify classification and prioritization for QoS across the internet. This means that inbound prioritization of traffic to your site is not possible. QoS is only effective when traffic is leaving your site. So, in order to ensure high quality voice on inbound voice traffic, there must always be more inbound bandwidth available then that is being consumed. Additional, for voice traffic, classification and prioritization should be used to ensure that voice traffic is prioritized higher then all other traffic.
Outbound or "upload" bandwidth is often the lower of the two quoted bandwidth numbers when looking at broadband services. Upload bandwidth for voice calls is the same as download bandwidth, 100kbps per call. When sizing the upload bandwidth, two options are available. First you can simply ensure there is enough bandwidth available so that the upload bandwidth is never fully consumed. Second, if the full bandwidth is consumed at times, ensure that proper Outbound QoS is in place to ensure that when a fight for bandwidth takes place, the voice services will always be passed without delays or drops. The best case scenario is a combination of both ensuring that for your normal network load enough bandwidth is available, and outbound QoS is in place in the event of any intermittent changes in outbound usage.
Voice Traffic Delivery
- Packet Loss: Less then 0.1%
- Round Trip Delay (Ping): Less then 100ms
- Jitter: Less then 20ms
- Bi-Directional Bandwidth Per Call Path: ~100kbps
Outbound Voice Classification and Marking
- UniVoIP phones will automatically mark Voice traffic with DSCP 46, also known as EF or Expedite Forward.
- Voice Signaling traffic will be marked with DSCP 26 or DSCP 24, also known as AF31 and CS3.
- Your router or firewall should be configured to recognize these markings, and queue the traffic for Prioritization. If using a managed or “Smart” L2 or L3 switch, it is important to verify that the switch is not stripping the L3 priority values from traffic. It is common for switches to do this by default, and should be reviewed to ensure voice and signing markings will not be removed.
Outbound Voice Prioritization
- Voice traffic when being sent outbound by your router or firewall towards your ISP should always be put into a Low Latency Queue. The size of this prioritization queue should be sized according to your maximum expected call count, multiplied by 100kbits. Voice Signaling traffic should be put into a bandwidth reserved queue that is roughly 10% the size of the Low Latency Voice Queue.
- Priority treatment can be given to all traffic associated with the UniVoIP Service Edge Networks. This type of prioritization would be implemented at the firewall or router managing the ISP connection.