---
title: "G.722 vs Opus for Voice AI in 2026: The Real Tradeoff Beyond Bitrate"
description: "G.722 fixed 64 kbps with sub-2 ms latency or Opus 6-510 kbps with adaptive bitrate? For voice AI agents the answer depends on whether your media path crosses a SIP trunk or stays end-to-end WebRTC."
canonical: https://callsphere.ai/blog/vw3d-g722-vs-opus-codec-tradeoffs-2026
category: "AI Infrastructure"
tags: ["Opus", "G.722", "Codec", "WebRTC", "Voice AI"]
author: "CallSphere Team"
published: 2026-03-17T00:00:00.000Z
updated: 2026-05-07T09:59:38.189Z
---

# G.722 vs Opus for Voice AI in 2026: The Real Tradeoff Beyond Bitrate

> G.722 fixed 64 kbps with sub-2 ms latency or Opus 6-510 kbps with adaptive bitrate? For voice AI agents the answer depends on whether your media path crosses a SIP trunk or stays end-to-end WebRTC.

> Opus wins almost every codec benchmark on the public internet, but G.722 still wins the SIP trunk negotiation 90% of the time. For an AI voice builder in 2026, picking between them is less a quality argument than an architecture argument.

## Background

```mermaid
flowchart TD
  Out[Outbound campaign] --> Twilio[Twilio Voice API]
  Twilio --> STIR[STIR/SHAKEN attestation]
  STIR --> Carrier[Originating carrier]
  Carrier --> Term[Terminating carrier]
  Term --> Recipient[Recipient phone]
  Recipient --> Webhook[/voice webhook/]
  Webhook --> Agent[AI sales agent]
```

CallSphere reference architecture

G.722 was standardized by ITU-T in 1988 and is the original HD voice codec: 7 kHz audio band over a 16 kHz sample rate at a fixed 64 kbps, sub-2 ms algorithmic delay. It is the de-facto interop codec for SIP wideband. Opus was standardized as RFC 6716 in 2012 by IETF, supports 6 kbps to 510 kbps variable bitrate, sample rates from 8 to 48 kHz, and is mandatory-to-implement in WebRTC.

For voice AI specifically, both are reasonable choices. Opus surpasses G.722 on raw audio quality, especially below 32 kbps, and adapts to network conditions. G.722 is rock-stable, ubiquitous, and adds essentially zero CPU on the encode/decode path.

## Technical deep-dive

The key tradeoffs:

| Aspect | G.722 | Opus |
| --- | --- | --- |
| Bitrate | Fixed 64 kbps | 6-510 kbps adaptive |
| Sample rate | 16 kHz | 8/12/16/24/48 kHz |
| Algorithmic delay | 2%, prefer Opus where available.
6. Run an ASR A/B test: same calls transcribed via G.722 path and Opus path, compare WER on names and digits.
7. Avoid letting the SIP gateway transcode mid-call; either pick a codec the AI bridge can accept natively or transcode once at the edge.

## FAQ

**Should I force Opus on every leg if I can?**
On WebRTC legs, yes. On SIP legs, only if your trunk and your AI bridge both support it; otherwise you add an unnecessary transcode.

**Does Opus's variable bitrate confuse the model?**
No. The model receives PCM samples after decode, not the codec frames. Bitrate variation only affects what the network carries.

**Is G.722 still patent-encumbered?**
The original patents expired long ago. It is freely implementable and ubiquitous in open source (FreeSWITCH, Asterisk, PJSIP).

**What about G.729 or AMR-WB?**
G.729 is narrowband 8 kHz, worse for ASR than G.722, no advantage for AI. AMR-WB is similar quality to G.722 but more common on mobile interconnect than SIP trunks.

**How much CPU do these add?**
G.722 is essentially free (sub-1% of a vCPU per call). Opus at 48 kHz with FEC is 2-5% per call. Negligible at our concurrency until you cross a few thousand simultaneous calls.

## Sources

- [Telnyx: How Opus and G.722 codecs turbocharge AI](https://telnyx.com/resources/voice-ai-hd-codecs)
- [Opus Codec official comparison](https://www.opus-codec.org/comparison/)
- [Connection Technologies: VoIP Audio Codecs Explained](https://connection-technologies.co.uk/help/voip-sip-trunking/sip-trunk-codecs-explained)

Start a [14-day trial](/trial) to test the codec path live, see [pricing](/pricing), or [contact us](/contact) about wideband AI voice negotiation.

---

Source: https://callsphere.ai/blog/vw3d-g722-vs-opus-codec-tradeoffs-2026