VoIP and SIP trunk bandwidth calculator
Calculate the bandwidth your SIP trunk or IP phones need per call and in total.
Calculated in your browser. Nothing is sent to our servers.
Choose a codec, packet interval and number of simultaneous calls to see the real bandwidth a SIP trunk or IP phone system needs, including IP, UDP, RTP and link-layer headers. Add VPN overhead if calls run over an IPsec tunnel between branches.
Why a 64 kbps codec needs 87 kbps
G.711 produces 64 kbps of voice, but every 20 ms packet also carries 40 bytes of IP, UDP and RTP headers plus 18 bytes of Ethernet framing. At 50 packets per second that adds up to 87.2 kbps per call in each direction. G.729 compresses voice to 8 kbps, but the same headers bring it to 31.2 kbps.
Choosing a codec
- G.711: best quality and no transcoding with the PSTN. Use it on LANs and fast links.
- G.729: a third of the bandwidth, good for small branch links.
- G.722: HD voice at the same bandwidth as G.711.
- Opus: adaptive, used by WebRTC and softphones.
Quality matters more than bandwidth
Voice needs low latency (under 150 ms one-way), low jitter and almost no packet loss. Mark voice traffic with DSCP EF and give it priority queuing on WAN links, or call quality will suffer even when the bandwidth is sufficient.
Common questions
How much bandwidth does one VoIP call use?
About 87 kbps each way for G.711 and 31 kbps for G.729 at 20 ms packets over Ethernet. The calculator adds headers for your settings.
Is the bandwidth the same for upload and download?
Yes. A call sends and receives the same amount, so size both directions of the link.
How many concurrent calls do we need?
As a rule of thumb, 1 trunk channel for every 4–6 users in an office. For a precise answer from your call volume, use the Erlang B calculator.
