How to Use the VoIP Codec Packet Bandwidth Calculator
-
1
Select a VoIP audio codec: G.711 (uncompressed 64 kbps), G.729 (compressed 8 kbps), G.722 (wideband HD), or Opus.
-
2
Set packetization interval (typically 20 ms, producing 50 packets per second).
-
3
Specify number of concurrent calls (e.g. 1, 10, 50, 100).
-
4
Review total bandwidth consumption in kbps and Mbps including Ethernet L2, IP, UDP, and RTP header overhead.
Practical Example & Use Case
Common Real-World Scenario
Calculating that a 20-call SIP trunk using G.711u (64kbps audio + 23.2kbps packet overhead) requires 1.74 Mbps dedicated upload and download bandwidth.
Technical Notes & Limitations
Browser Engine Architecture
Calculates standard Layer 2 Ethernet frame overhead (87.2 kbps per G.711 call at 20ms ptime).
Frequently Asked Questions
Why is VoIP bandwidth higher than the codec bitrate?
Every 20ms voice packet requires RTP (12 bytes), UDP (8 bytes), IP (20 bytes), and Ethernet (18 bytes) headers, adding significant protocol overhead.