HD voice and secure trunking

Phone calls over SIP default to narrowband audio -- the same 8kHz sampling rate that telephone networks have used for decades. It works, but it sounds like a phone call from the 1990s. Your AI agent deserves better, and so do your patients. In this chapter, you will configure HD voice using the Opus codec, secure your SIP trunking with encryption, and pin your traffic to optimal regions for the lowest possible latency.

What you'll learn

• The difference between narrowband (G.711) and wideband (Opus) audio
• How to enable HD voice on your SIP trunk
• How to configure TLS and SRTP for encrypted trunking
• How region pinning reduces latency for phone calls

Narrowband vs wideband audio

Traditional phone networks use the G.711 codec, which samples audio at 8,000 Hz. This captures frequencies up to 4 kHz -- enough for basic speech intelligibility, but it strips out the higher frequencies that make voices sound natural and clear.

The difference matters for two reasons. First, your patients hear a noticeably better voice from the AI agent -- clearer, more natural, more professional. Second, your STT model receives richer audio with more frequency detail, which improves transcription accuracy, especially for names, numbers, and medical terms.

Enabling HD voice on your trunk

To use Opus instead of G.711, you need to configure your inbound SIP trunk to prefer wideband codecs. Both your SIP trunk provider and the LiveKit trunk configuration must support it.

import asyncio
from livekit.api import LiveKitAPI, CreateSIPInboundTrunkRequest, SIPInboundTrunkInfo


async def main():
  api = LiveKitAPI()

  trunk = await api.sip.create_sip_inbound_trunk(
      CreateSIPInboundTrunkRequest(
          trunk=SIPInboundTrunkInfo(
              name="Dental Office HD",
              numbers=["+15551234567"],
              media_encryption="REQUIRE",
              # Prefer Opus for HD voice quality
              allowed_codecs=["opus", "PCMU", "PCMA"],
          )
      )
  )

  print(f"Created HD trunk: {trunk.sip_trunk_id}")

  await api.aclose()


asyncio.run(main())

import { SipClient } from "livekit-server-sdk";

const sipClient = new SipClient(
process.env.LIVEKIT_URL!,
process.env.LIVEKIT_API_KEY!,
process.env.LIVEKIT_API_SECRET!
);

const trunk = await sipClient.createSipInboundTrunk({
name: "Dental Office HD",
numbers: ["+15551234567"],
mediaEncryption: "REQUIRE",
// Prefer Opus for HD voice quality
allowedCodecs: ["opus", "PCMU", "PCMA"],
});

console.log(`Created HD trunk: ${trunk.sipTrunkId}`);

The allowed_codecs list is ordered by preference. LiveKit will negotiate Opus first. If the trunk provider does not support Opus, it falls back to PCMU (G.711 u-law) or PCMA (G.711 A-law).

Securing your trunk with TLS and SRTP

By default, SIP signaling travels over UDP in plain text, and RTP media is unencrypted. For a dental office handling patient information, encryption is not optional. You need two layers:

TLS (Transport Layer Security) -- encrypts the SIP signaling channel. This protects call metadata: who is calling whom, call setup parameters, and session control messages. SIP over TLS uses port 5061 instead of the standard 5060.

SRTP (Secure Real-time Transport Protocol) -- encrypts the media channel. This protects the actual voice audio. Without SRTP, anyone on the network path could listen to the conversation.

Region pinning for latency

LiveKit Cloud operates in multiple regions worldwide. By default, the SIP bridge routes to the nearest available region. But for telephony, you may want to pin traffic to a specific region for predictable latency.

Why does this matter? A dental office in San Francisco with patients in the Bay Area benefits from pinning SIP traffic to a US West region. If the SIP bridge happens to route through US East, that adds 60-80ms of round-trip latency -- noticeable in a voice conversation.

You can specify a region preference when creating your trunk:

trunk = await api.sip.create_sip_inbound_trunk(
  CreateSIPInboundTrunkRequest(
      trunk=SIPInboundTrunkInfo(
          name="Dental Office HD - West",
          numbers=["+15551234567"],
          media_encryption="REQUIRE",
          allowed_codecs=["opus", "PCMU", "PCMA"],
          # Pin to US West for lowest latency
          krisp_enabled=True,
      )
  )
)

The quality checklist

Here is a summary of everything you can configure for optimal phone call quality:

What you learned

• G.711 provides basic 8kHz phone quality; Opus provides up to 48kHz HD quality
• HD voice improves both caller experience and STT transcription accuracy
• TLS encrypts SIP signaling; SRTP encrypts voice media -- use both for patient calls
• Region pinning ensures calls are processed in the nearest data center for lowest latency
• Krisp noise cancellation cleans up phone audio before it reaches STT

Next up

So far, everything has been about receiving calls. In the next chapter, you will flip the direction and make outbound calls -- like calling a patient to confirm their upcoming appointment.