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VoIP Bandwidth Calculator: Sizing Your Network for Voice
Why Bandwidth Calculation Matters
Insufficient bandwidth is the most common cause of VoIP quality problems. When voice traffic exceeds available network capacity, packets queue, delay, and drop - manifesting as choppy audio, echo, and dropped calls. Proper bandwidth calculation before deployment prevents these issues and ensures call quality meets business standards.
Calculating VoIP bandwidth involves three factors: number of concurrent calls, codec selection, and protocol overhead. Understanding these factors enables accurate network sizing and prevents the costly discovery of capacity problems after deployment.
Codec Bandwidth Requirements
Each codec has specific bandwidth requirements for the voice payload. G.711 uses 64 kbps per call direction, meaning a single call requires 64 kbps in each direction (128 kbps total for bidirectional communication). G.729 uses 8 kbps per direction, requiring 16 kbps total per call.
However, raw codec bandwidth isn't the total requirement. IP packets carrying voice include headers from RTP, UDP, and IP protocols. These headers add approximately 54 bytes per packet, significantly increasing actual bandwidth consumption beyond the codec rate.
Protocol Overhead and Total Bandwidth
With G.711, packets are typically sent every 20 milliseconds, meaning 50 packets per second. Each packet carries 160 bytes of voice data plus 54 bytes of headers, totaling 214 bytes per packet. At 50 packets per second, this equals 107,000 bytes per second, or approximately 856 kbps - significantly more than the raw 64 kbps codec rate.
For G.729, packets also sent every 20ms carry 20 bytes of voice data plus 54 bytes of headers, totaling 74 bytes per packet. At 50 packets per second, this equals 29,600 bytes per second, or approximately 237 kbps per call - compared to the raw 8 kbps codec rate.
Compressed RTP (cRTP) on WAN links can reduce header overhead by compressing IP/UDP/RTP headers. With cRTP, the 54-byte header compresses to approximately 2-4 bytes, dramatically reducing total bandwidth requirements on expensive WAN links. However, cRTP requires router configuration and processing resources.
Calculating Total Requirements
To calculate total bandwidth for a deployment: multiply concurrent calls by per-call bandwidth (including overhead). For 20 concurrent calls using G.711: 20 x 856 kbps = 17,120 kbps (approximately 17 Mbps). For 20 calls using G.729: 20 x 237 kbps = 4,740 kbps (approximately 5 Mbps).
Add 20-30% headroom for traffic variations, retransmissions, and signaling overhead. For 20 G.711 calls: 17 Mbps x 1.3 = approximately 22 Mbps. For 20 G.729 calls: 5 Mbps x 1.3 = approximately 6.5 Mbps. This headroom ensures quality during peak usage and accommodates growth.
Network Configuration for Voice Quality
Beyond raw bandwidth, network configuration affects voice quality. Quality of Service (QoS) settings prioritize voice traffic over data, ensuring voice packets receive preferential treatment during congestion. Without QoS, even networks with sufficient bandwidth can experience quality problems during large file transfers or other data bursts.
Jitter buffers in VoIP endpoints compensate for minor timing variations in packet arrival. Networks with variable latency (jitter) need larger jitter buffers, adding delay to conversations. Networks with consistent latency can use smaller buffers, minimizing delay. Network design should minimize jitter through proper switch and router configuration.
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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