VoIP bandwidth calculator

Choose your codecs, describe the network, and enter your call load. The result updates as you type, and every number can be traced step by step below.

Codecs

Network

Header details

IP version VLAN tags4 B each. 2 = QinQ. SRTP authentication tagSRTP authentication-tag bytes added to each packet. SRTP MKI lengthOptional Master Key Identifier per packet; counted only when SRTP is selected. RTP CSRC entries4 B each. Only mixers add these; pass-through SBC media is 0. RTP extension wordsRFC 8285, mostly WebRTC. SIP trunks use 0. Extra bytes per packetTunnel or IPsec overhead.

Always included: UDP 8 B, Ethernet header 14 B, FCS 4 B, padding to a 64 B minimum frame. On-wire adds 20 B for preamble, start delimiter and inter-frame gap. SRTP adds the selected authentication tag and optional MKI; the MKI is ignored for plain RTP. BGP and plain L2/L3 routing add no packet bytes; for GRE, IPsec or MPLS enter the overhead as extra bytes.

Load

Concurrent callsActive at the same moment. Drives media bandwidth. Calls per secondNew calls per second. Drives signalling and CPU. Call duration (s)Calls = CPS × duration. Edit any two; the third is calculated. Measure at Media passes the deviceTwice gives total SBC throughput. Talking time (%)100 = no silence suppression. RTCP Safety margin

VoIP bandwidth calculator

Codecs

Network

Load

Where the bytes go

Per-codec figures

Working

Scaling with call count