Cloud & Telecom Solutions

Supported codecs Print

  • 42

VoIP audio codecs — Full comparison

The audio codec determines the quality and the bandwidth consumed by each call. WebTic supports the following codecs:

Comparison table

CodecBandwidthMOS scoreAudio bandwidthRecommended use
G.711 PCMU/PCMA64 kbps4.1 – 4.33.5 kHzUniversal standard, maximum quality
G.7298 – 12 kbps3.9 – 4.03.5 kHzLow-bandwidth links
G.72248 – 56 kbps4.1 – 4.37 kHz (HD)HD calls, premium quality
Opus6 – 40 kbps4.2 – 4.520 kHzWebRTC, adaptive, modern

Bandwidth calculation

Formula: Simultaneous calls × Codec bandwidth = Total bandwidth

  • 100 G.711 calls: 100 × 64 kbps = 6.4 Mbps
  • 100 G.729 calls: 100 × 10 kbps = 1 Mbps (84% savings)
  • 100 G.722 calls: 100 × 56 kbps = 5.6 Mbps

G.711 — The reference standard

G.711 (PCMU = µ-law for North America, PCMA = A-law for Europe) is the most widespread codec. It does not compress the audio, preserving quality with no processing latency. Recommended for WebTic trunks.

G.729 — For limited connections

Compresses the audio using 20 ms packets. Reduces bandwidth by 87% compared with G.711. Note: a G.729 license may be required on some platforms. Quality is slightly lower (artefacts on fax and DTMF).

G.722 — HD calls

Wideband codec (7 kHz) offering markedly better voice quality. Ideal for contact centers and conferences. Compatible with 3CX, Cisco, Yealink.

Recommended priority order (SDP)

m=audio 5004 RTP/AVP 8 0 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000

Note: Fax must always use G.711 or T.38 — never G.729.


Was this answer helpful?

« Back