Codec Selection for Outbound Call Centers: G.711, G.729, and Opus Compared

What a Codec Actually Does

A codec (coder-decoder) compresses raw PCM audio into a payload that fits inside an RTP packet, then decompresses it at the other end. Every codec makes a trade: more compression means smaller packets but more processing delay and potential quality loss. In outbound call centers, you care about three things: audio fidelity (does the contact hear you clearly), bandwidth per channel, and transcoding cost when your codec hits a PSTN gateway running something different.

G.711: The Baseline

G.711 (u-law in North America, a-law in Europe) is uncompressed PCM at 64 kbps. It is the native format of the PSTN and introduces no additional codec delay beyond packetization — typically 20 ms packets with a 5 ms tail. Total one-way codec latency: approximately 25 ms.

Bandwidth per channel: 87 kbps including IP/UDP/RTP headers on a 20 ms packetization interval. For 100 simultaneous outbound calls, that is 8.7 Mbps of committed uplink.

Because G.711 matches PSTN's native encoding, any PSTN gateway on the terminating side performs no transcoding. This matters for outbound call quality: every transcoding step adds approximately 15–30 ms of latency and introduces quantization artifacts. A G.711 → G.711 path from your SBC to the far-end PSTN gateway is bit-for-bit lossless.

When to use G.711: your WAN has ample symmetric bandwidth, you have a flat-rate trunk (so bandwidth cost is absorbed in the seat fee rather than per-minute metering), and you want maximum compatibility with any downstream termination carrier. This is the default on UnlimCall's SIP trunk and the recommended starting point.

G.729: The Bandwidth Saver

G.729 compresses 8 kHz PCM audio to 8 kbps using ACELP (Algebraic Code-Excited Linear Prediction). With headers, each channel consumes approximately 32 kbps — less than 40% of G.711. For a 100-channel floor, that drops from 8.7 Mbps to 3.2 Mbps.

The cost: codec delay. G.729 uses a 10 ms frame size but requires a look-ahead buffer, adding 5 ms of algorithmic delay. Combined with packetization (20 ms intervals), one-way codec latency is approximately 35 ms. More significant for call quality is the quality ceiling. G.729 is a narrowband codec (300–3400 Hz), identical to G.711 in frequency response, but the lossy compression produces audible artifacts on noisy environments — a contact in a busy office or on a mobile will sound worse than on G.711.

G.729 also introduces transcoding cost. If your SBC sends G.729 and the PSTN gateway on the terminating carrier only accepts G.711 (common), every call transits a transcoder — adding latency, complexity, and a potential quality step-down.

When to use G.729: your WAN uplink is genuinely constrained (under 5 Mbps for a 50-seat floor) and you cannot upgrade to symmetric fiber. For most outbound operations that have met the network requirements baseline, G.711 is the correct choice.

Opus: Wideband Audio for IP-to-IP Paths

Opus is a variable-bitrate wideband codec covering 8–48 kHz, standardized in RFC 6716. At 24 kbps with 20 ms frames, it delivers noticeably better speech intelligibility than G.711 — especially on mobile handsets where the far-end subscriber has a wideband-capable device.

The catch for outbound call centers: Opus only helps on the IP segment. The moment your call hits the PSTN, it is transcoded to G.711 or G.729 at the gateway. For outbound calls terminating to landlines and mobile phones via PSTN interconnect — which is every call on UnlimCall's 33-market network — the contact never hears the Opus-encoded signal.

Opus is appropriate for internal agent monitoring, supervisor barge-in over WebRTC, or softphone-to-softphone paths that never touch the PSTN. For the actual outbound leg, G.711 delivers the same quality with zero transcoding overhead and maximum compatibility.

Codec Negotiation in SDP: What to Inspect

When your SBC initiates a call, it sends an SDP offer listing supported codecs in priority order. The far-end SBC picks the first match. If you list G.711 first and the terminating gateway supports it, no transcoding occurs. If you list G.729 first, the gateway may accept it even if it would prefer G.711, forcing an unnecessary transcoding on downstream re-origination.

A clean SDP offer for outbound:

`` m=audio 16384 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ``

Payload 0 (PCMU/G.711 u-law) is first priority. Payload 8 (PCMA/G.711 a-law) is second for European PSTN gateways. G.729 is listed as a fallback. DTMF telephone-event (RFC 2833) is always included to avoid in-band tone detection failures.

Configure your preferred codec order in your SBC's SIP profile. For Asterisk: allow=ulaw&alaw&g729 in sip.conf. For FreeSWITCH, set <param name="inbound-codec-prefs" value="PCMU,PCMA,G729"/> in the Sofia profile.