VoIP.info Calculator

Codec bandwidth comparison

Compare G.711, G.722, G.729 and an editable Opus bitrate under the same packet assumptions.

Reviewed 2026-09-07Intermediate
Estimated IP traffic · each direction
One RTP stream per call on the measured link
CodecPayload kbit/sPer call kbit/sAll calls Mbit/s
G.71164800.8
G.722 (64 kbit/s mode)64800.8
G.729 (8 kbit/s mode)8240.24
Opus (entered bitrate)32480.48

Includes IP + UDP + basic RTP headers and the entered extra bytes. No silence suppression, retransmission or planning allowance. G.722 and G.711 have different audio bandwidth despite equal traffic in the selected modes. Check which packetization intervals both peers support.

Compare a defined stream

The table models one RTP audio stream per call, in each direction on the measured link. G.711 and the selected G.722 mode both carry 64 kbit/s of audio payload, although G.722 represents wider-band audio. G.729 is modeled at its 8 kbit/s mode. Opus bitrate is editable because its actual output depends on encoder settings and content.

An equal traffic estimate is not a quality ranking. Codec availability, licensing where relevant, endpoint support, transcoding and packet-loss behavior also matter. Confirm the actual negotiated codec and packet duration in a capture.

Count headers and packets

Packets per second = 1,000 ÷ packet duration in milliseconds. Basic RTP contributes 12 bytes, UDP 8 bytes and IPv4 or IPv6 contributes 20 or 40 bytes before optional headers. Extra bytes per packet can represent a specific measured addition, but are zero by default.

Per-direction kbit/s = payload kbit/s + total header bytes × 8 ÷ packet duration in milliseconds. At 20 ms with IPv4 and no extra bytes, G.711 and G.722 are 80 kbit/s, G.729 is 24 kbit/s and 32 kbit/s Opus is 48 kbit/s. Ten G.711 calls therefore use 0.8 Mbit/s in each direction before other overhead or allowance.

Know what is outside the estimate

The calculation excludes Ethernet framing, VPN encapsulation, SRTP expansion, RTP extensions and RTCP unless you explicitly account for the relevant bytes. It also excludes retransmissions, redundancy, silence suppression and variable packet sizes. Not every codec or endpoint supports every modeled packetization interval.

A longer interval reduces header traffic but adds packetization delay and makes each lost packet cover more audio. Use the bandwidth calculator for a planning allowance, and evaluate delay, jitter and loss separately.

Sources & applicability

Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.