Bandwidth Requirements for SIP Trunking: Calculate Your Needs
📡 Summary: Getting SIP trunk bandwidth right prevents jitter, call drops, and poor audio. This guide covers codec bitrates (G.711, G.729, opus), per-call overhead, formulas to calculate for concurrent calls, and real-world examples. Learn how to size your internet connection for flawless voice quality and cost-efficient trunking.
📖 Table of Contents
SIP trunking replaces traditional phone lines with an IP-based connection, but its success depends entirely on stable, sufficient bandwidth. Unlike web browsing or email, voice traffic is extremely sensitive to delay, packet loss, and jitter. Under-provisioning your internet link leads to choppy audio, dropped calls, and frustrated customers.
Many businesses overestimate or underestimate their needs. A single SIP trunk channel can consume anywhere from 30 kbps to 100+ kbps depending on the codec, packetization interval, and layer-2 overhead. To calculate accurately, you must consider concurrent call volume, codec selection, and network overhead (Ethernet/IP/UDP/RTP). This guide gives you the exact formulas and tools to size your connection like a pro.
Whether you run a 5-line dental office or a 50-seat call center, you'll learn how to compute required upload/download speeds, reserve headroom, and prioritize voice traffic using QoS. Let's break down the numbers.
🎙️ Voice Codecs & Their Real Bandwidth Consumption
Codecs compress audio. The most common for SIP trunking are G.711 (uncompressed, high quality) and G.729 (compressed, lower bandwidth). Each codec has a payload bitrate, but adding IP/UDP/RTP headers and Ethernet overhead increases total usage by ~20–30%.
📊 Bandwidth per call (including all overheads)
| Codec | Bitrate (kbps) payload | Total with overhead (kbps) | Audio Quality | Use case |
|---|---|---|---|---|
| G.711 (ulaw/alaw) | 64 kbps | ~87–100 kbps | Toll-quality | LAN, fiber, unlimited bandwidth |
| G.729A | 8 kbps | ~31–40 kbps | Good (slight compression) | Low bandwidth links, WAN |
| G.722 (HD Voice) | 64 kbps (wideband) | ~100–114 kbps | Excellent HD | High-speed, modern UC |
| Opus (adaptive) | 20–64 kbps | ~40–100 kbps | Dynamic | Flexible, WebRTC |
🔍 Key insight: Overhead includes Ethernet (14 bytes), IP (20), UDP (8), RTP (12) and frame overhead. Using a 20ms packetization interval (default for most PBXs) gives the values above. Longer intervals reduce overhead but increase latency.
🧮 Bandwidth Calculation Formula (Simple & Accurate)
To find total bandwidth needed for your SIP trunk, use this formula:
Total Bandwidth (kbps) = (Bandwidth per call in kbps) × (Number of concurrent calls) × 1.2 (safety margin)
✅ Example: 10 concurrent calls with G.711 codec (95 kbps each) → 10 × 95 = 950 kbps → +20% margin = 1,140 kbps (~1.14 Mbps) each for upload AND download.
✅ Example (G.729): 15 calls × 36 kbps = 540 kbps → +20% = 648 kbps (0.65 Mbps) symmetric.
Note: SIP trunking requires bidirectional bandwidth — one stream for transmit (your voice) and one for receive (far end). So your internet plan's upload speed must match the requirement as well as download. Asymmetric connections (e.g., 200/20 Mbps) might have plenty of download but insufficient upload for many calls.
📋 Bandwidth Tables: From 1 to 50 Concurrent Calls
Recommended bandwidth (Mbps) — includes 20% safety margin
| Concurrent Calls | G.711 (Mbps) | G.729 (Mbps) | G.722 HD (Mbps) |
|---|---|---|---|
| 1 call | 0.10 Mbps | 0.04 Mbps | 0.12 Mbps |
| 5 calls | 0.50 Mbps | 0.20 Mbps | 0.60 Mbps |
| 10 calls | 1.00 Mbps | 0.40 Mbps | 1.20 Mbps |
| 20 calls | 2.00 Mbps | 0.80 Mbps | 2.40 Mbps |
| 30 calls | 3.00 Mbps | 1.20 Mbps | 3.60 Mbps |
| 50 calls | 5.00 Mbps | 2.00 Mbps | 6.00 Mbps |
💡 Pro tip: Always reserve additional bandwidth for data traffic (email, browsing, cloud apps). For mission-critical voice, dedicate a VLAN and implement QoS.
📈 Visual: Bandwidth Growth per Concurrent Call (G.711)
Chart shows required bandwidth per direction (up/down) using G.711 with overhead and margin. For G.729, divide numbers by ~2.5.
⚙️ Best Practices: QoS, Jitter Buffer & Headroom
🔧 QoS Configuration
Set DSCP EF (46) for SIP/RTP traffic on your router. Prioritize voice over all other packets to avoid latency spikes. Use bandwidth limiting for non-critical apps.
📏 Jitter Buffer
Most IP-PBXs use adaptive jitter buffers (40–120ms). Increase buffer only if you have unstable connections; otherwise keep default to avoid delay.
🚦 Over-Provisioning
Add 20–30% extra headroom above peak calculated usage. Sudden bursts (conference calls, call storms) can degrade quality without cushion.
Also monitor your SIP trunk with tools like PRTG or Wireshark. Latency should stay below 150ms, jitter < 30ms, packet loss < 1% for acceptable call quality.
🚀 Optimize your SIP trunk bandwidth with NetviaVoice
We provide end-to-end SIP trunking, codec tuning, and managed QoS. Get a custom bandwidth calculator for your business.
Visit Homepage❓ 5 FAQs on SIP Trunk Bandwidth (Google & LLM Verified)
With G.711 codec (uncompressed), one call uses about 87–100 kbps including all overhead. With G.729 compression, ~31–40 kbps per call. Use the higher value for safety.
Insufficient bandwidth leads to jitter, packet loss, robotic audio, call drops, and poor voice quality. QoS settings and adequate headroom (20-30%) are critical.
G.729 or G.723.1 reduce bandwidth by ~70% compared to G.711, but at the cost of slightly lower audio fidelity. G.711 is recommended for LAN or high-speed connections.
Multiply bandwidth per call (e.g., 100 kbps for G.711) by number of calls (10) = 1000 kbps (1 Mbps) each for upload and download. Add 20% margin → 1.2 Mbps.
Yes, SIP trunking requires both upload and download capacity because voice is bidirectional. Asymmetric connections (e.g., 100/10 Mbps) can work if upload meets demand.
📢 Speak to a SIP engineer — we’ll calculate your exact bandwidth requirement and recommend the best trunk size.