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Quality of Service Configuration for Optimal VoIP Performance
Why QoS Matters for VoIP
VoIP packets traveling through data networks compete for bandwidth with other traffic types including file transfers, video streaming, web browsing, and cloud application data. Without Quality of Service controls, large data transfers can overwhelm network links, causing delayed or dropped voice packets that listeners perceive as choppy audio, one-way conversations, or dropped calls.
QoS mechanisms classify different traffic types and apply priority rules ensuring time-sensitive voice packets receive preferential treatment. Since even small delays (150 milliseconds or more) become noticeable during live conversation, proper QoS implementation is foundational for acceptable VoIP quality on shared networks.
QoS Classification and Marking
QoS begins with identifying voice packets and marking them for prioritization. Differentiated Services Code Point values in IP packet headers signal network equipment how to treat each packet. Voice signaling and voice media traffic receive dedicated DSCP values - typically EF (Expedited Forwarding) for media and CS3 or AF31 for signaling - that network devices recognize throughout the transmission path.
Classification can occur at IP phones, network switches, or routers depending on network design. Best practice has trusted devices - phones or managed switches - apply DSCP markings at the network edge, ensuring voice receives appropriate priority from the moment packets enter the network.
Queuing and Scheduling Mechanisms
Once packets are classified and marked, network devices use queuing algorithms to prioritize higher-priority traffic during congestion. Strict Priority Queuing reserves dedicated bandwidth for voice packets, transmitting them before any lower-priority traffic. Weighted Fair Queuing and Class-Based Weighted Fair Queuing allocate bandwidth proportionally while ensuring priority classes receive their guaranteed share.
Proper bandwidth allocation for voice means reserving enough capacity for maximum concurrent calls plus overhead. As a rule of thumb, 100 kbps per concurrent call direction (upstream and downstream) for G.711 codec ensures sufficient bandwidth even with protocol overhead. Networks should be engineered so reserved voice capacity plus expected data usage doesn't exceed total available link capacity.
End-to-End QoS Considerations
QoS effectiveness depends on consistent implementation across the entire network path. If one hop along the route lacks proper QoS configuration, voice quality suffers at that point regardless of other correctly configured segments. For multi-site organizations using WAN links, QoS across provider networks (purchased as Class of Service or MPLS with QoS) represents an often-overlooked component needing verification during service procurement.
Network jitter buffers in VoIP endpoints compensate for minor packet arrival timing variations, but they cannot overcome sustained congestion without QoS. Monitoring tools measuring jitter, packet loss, and latency across network segments help identify where QoS policies require adjustment to maintain consistent call quality.
Redstone Systems, Inc. founded in Delaware, USA in December 2002, has been the ODM vendor for many well-known communications companies, serving the Southeast Asian market. In 2020, Redstone Systems will begin to return to the North America market with its self-developed brand.
Redstone has a complete product line of intelligent voice gateways, providing IP-PBXs, analog VoIP gateways (FXS/FXO), digital VoIP gateways (E1/T1), border appliances, and session boundary controllers (SBCs).
With advanced technology in digital signal processor (DSP), speech coding and speech processing, as well as efficient operational tools such as cloud remote management, auto provisioning, Redstone gateways are widely used in markets of enterprise communications, cloud communications, call centers, operators’ IMS/SIP trunks, bringing users friendly, efficient and reliable communication experience.
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