HD Voice for LiveKit Agents on Telnyx

HD Voice is now available on LiveKit on Telnyx across all four regions (SFO3, NYC1, ATL1, and SYD1), with G.722 and Opus wideband codecs for SIP audio. On LiveKit Cloud, HD Voice requires routing through a third-party SIP carrier. On Telnyx, wideband codecs run on the carrier network where your STT, TTS, and LLM inference already live. No external provider, no custom integration, no code changes.

Higher-fidelity audio means clearer conversations and less listener fatigue on long calls. It also feeds cleaner signal into colocated STT models, improving transcription accuracy for voice AI workflows.

What's New

• G.722 codec support: Wideband audio (7 kHz) works out of the box on every LiveKit region. If your SIP endpoint supports G.722, you get HD audio automatically. No trunk reconfiguration, no extra provider.
• Opus codec support: Full-band audio up to 48 kHz when SRTP is enabled on your SIP trunk. Opus is among the highest-fidelity codecs available for real-time voice, with adaptive bitrate that handles network conditions gracefully.
• All regions live: SFO3, NYC1, ATL1, and SYD1 are all running both codecs. Route calls to whichever region is closest to your SIP infrastructure for the lowest latency.

Why It Matters

• Enterprise SIP, built in: Wideband codecs are part of the carrier stack, not a bolt-on. Telnyx handles STIR/SHAKEN attestation, AMR-WB, G.722, Opus, and custom trunk configurations natively. No third-party SIP provider required.
• Sharper conversations: Wideband audio doubles the frequency range of standard narrowband calls, making voices clearer and reducing listener fatigue on long calls.
• Better AI pipeline input: Higher-fidelity audio feeds cleaner signal into colocated STT models, improving transcription accuracy for voice AI workflows.
• No trade-offs: G.722 is zero-config; Opus gives you top-tier quality when you need it.
• Consistent across regions: Every LiveKit region supports both codecs, so there's no feature gap regardless of where your traffic lands.

Getting Started

1. G.722: Ensure your SIP endpoint advertises G.722 in its SDP offer. Most modern endpoints do this by default, and LiveKit will negotiate it automatically.
2. Opus: Enable SRTP on your SIP trunk configuration. Once SRTP is active, LiveKit will negotiate Opus when the endpoint supports it.
3. Route your SIP traffic to any LiveKit region (SFO3, NYC1, ATL1, or SYD1).

Use G.722 for broad compatibility with existing SIP hardware, or Opus for maximum fidelity when SRTP is feasible. See the LiveKit Telephony documentation for full configuration details.