How much bandwidth does VoIP use? It depends on the codec, the number of concurrent calls, and your network overhead. This guide gives you the exact numbers and a simple formula to calculate your requirements.
For background on the codec choices below — quality, compatibility, and use cases — see VoIP codec comparison.
| Codec | Payload kbps | With overhead | Notes |
|---|---|---|---|
| G.711 (PCMU/PCMA) | 64 kbps | 87 kbps | Standard quality, no compression |
| G.722 | 64 kbps | 87 kbps | Wideband HD audio, same bandwidth as G.711 |
| G.729 | 8 kbps | 32 kbps | Compressed, lower quality under loss |
| G.723.1 (6.3k) | 6.3 kbps | 20 kbps | Very compressed, noticeable quality reduction |
| Opus (48kHz) | 6-510 kbps | varies | Adaptive, excellent quality at 32+ kbps |
| iLBC | 13.3 kbps | 28 kbps | Good loss resilience |
The "with overhead" column is what you actually need to plan for. IP header (20 bytes) + UDP header (8 bytes) + RTP header (12 bytes) = 40 bytes overhead per packet. With default 20ms ptime (50 packets/second), that's 50 x 40 x 8 = 16 kbps of overhead on top of the codec payload.
Formula: Concurrent calls x per-call bandwidth x 2 (bidirectional)
| Concurrent Calls | G.711 Required | G.729 Required |
|---|---|---|
| 5 calls | 870 kbps | 320 kbps |
| 10 calls | 1.74 Mbps | 640 kbps |
| 25 calls | 4.35 Mbps | 1.6 Mbps |
| 50 calls | 8.7 Mbps | 3.2 Mbps |
| 100 calls | 17.4 Mbps | 6.4 Mbps |
Add 20-30% headroom on top of calculated requirements for bursts, retransmissions, RTCP, and SIP signaling overhead. Don't plan for exactly 100% utilization — VoIP quality degrades sharply when bandwidth is fully saturated.
Every RTP packet carries 40 bytes of IP+UDP+RTP headers in addition to the audio payload. With IPv6, headers are 60 bytes. This overhead is significant at high packet rates:
cRTP compresses the 40-byte header to 2-4 bytes on point-to-point links. Reduces G.711 bandwidth from 87 kbps to ~68 kbps. Supported on Cisco routers and some SBCs. Only applies to the WAN link, not end-to-end.
WAN links are typically the bottleneck. Key planning principles:
Minimum recommended WAN bandwidth for reliable VoIP:
Switch from G.711 to G.729 — reduces per-call bandwidth from 87 kbps to 32 kbps (63% reduction). Trade-off: G.729 quality degrades faster under packet loss. Avoid G.729 on links with over 1% loss. Note that codec changes mid-network may require transcoding, which has its own CPU and quality costs.
Enable silence suppression / VAD — typical conversations have 50-60% silence. VAD stops sending RTP during silent periods, reducing average bandwidth by 40-50%. Trade-off: slight audio clipping at the start of speech, comfort noise required.
Increase ptime — using 30ms or 40ms ptime instead of 20ms reduces packet rate and therefore header overhead. Trade-off: higher latency and more audio lost per dropped packet.
Deploy cRTP on WAN links — compresses RTP headers on point-to-point WAN segments. Saves ~16 kbps per call on G.711 with no quality trade-off.
A G.711 VoIP call uses approximately 87 kbps per call per direction (64 kbps audio payload plus 23 kbps IP/UDP/RTP header overhead at 20ms ptime). G.729 uses approximately 32 kbps per call. For planning, multiply concurrent calls by per-call bandwidth by 2 (bidirectional) and add 25% headroom.
On a 10 Mbps symmetric connection with G.711 codecs, you can support approximately 50-55 concurrent calls while keeping voice bandwidth at 75% of link capacity (leaving headroom for data). With G.729, the same link can support 150+ concurrent calls. Always implement QoS to ensure voice traffic gets priority over data.
VoIP uses equal bandwidth in both upload and download directions — it is symmetric. Each call requires approximately 87 kbps upload AND 87 kbps download for G.711. This is important for asymmetric broadband connections where upload is significantly less than download — the upload side is usually the bottleneck.
Upload your PCAP to SIPSymposium. The analyzer measures per-call RTP bandwidth, detects packet loss from congestion, and identifies QoS marking issues.