> ## Documentation Index
> Fetch the complete documentation index at: https://dialnexa.com/docs/llms.txt
> Use this file to discover all available pages before exploring further.

# SIP Trunking In DialNexa

> Understand DialNexa SIP trunking, linking SIP credentials to a phone number, termination URI, optional credentials, nickname, SIP rate, routing, audio quality, and testing.

SIP trunking in DialNexa lets a workspace connect an existing phone number to a SIP trunk path while still using DialNexa agents, transcripts, recordings, call history, and monitoring.

<img src="https://mintcdn.com/dialnexa/O6bVvssz6DpTKOa0/images/documentation/screenshots/sip-trunk-modal.jpg?fit=max&auto=format&n=O6bVvssz6DpTKOa0&q=85&s=b7154c2cef02d3728b8aff715a28691d" alt="DialNexa SIP trunking modal showing per-minute SIP rate, phone number, termination URI, optional username, optional password, nickname, and save action." style={{ width: '100%', maxWidth: '1100px', margin: '8px 0 24px', border: '1px solid #e5e7eb', borderRadius: '6px' }} width="1710" height="985" data-path="images/documentation/screenshots/sip-trunk-modal.jpg" />

<Tip>
  SIP trunking gives you control over the telephony path. It also gives you the homework: prove the path works before live traffic.
</Tip>

## SIP Trunk Versus Plivo Number Purchase

| Option                | Use it when                                                              | Operational responsibility                                                                |
| --------------------- | ------------------------------------------------------------------------ | ----------------------------------------------------------------------------------------- |
| Plivo number purchase | You want to buy and route a number inside DialNexa.                      | Number availability, compliance, provider readiness, and assignment.                      |
| SIP trunking          | You already manage telephony and want DialNexa to connect to that route. | Termination URI, credentials, carrier behavior, caller ID, media path, and audio quality. |

## Fields In The SIP Trunk Modal

| Field               | Required     | Purpose                                               |
| ------------------- | ------------ | ----------------------------------------------------- |
| Phone Number        | Yes          | The E.164 number to link, such as `+12025551234`.     |
| Termination URI     | Yes          | The SIP destination URI used for routing.             |
| SIP Trunk User Name | Optional     | Username for trunks that require credentials.         |
| SIP Trunk Password  | Optional     | Password for trunks that require credentials.         |
| Nickname            | Optional     | Human-readable label for the number.                  |
| SIP rate            | Display only | Shows the per-minute SIP trunking rate before saving. |

## Link And Validate A SIP Number

<Steps>
  <Step title="Open Phone Numbers">
    Use the Phone Numbers page to start the SIP linking flow.
  </Step>

  <Step title="Enter the SIP details">
    Provide phone number, termination URI, and optional credentials if your trunk requires them.
  </Step>

  <Step title="Assign published versions">
    The number still needs inbound or outbound published agent versions. SIP decides the route, not the conversation behavior.
  </Step>

  <Step title="Place an inbound test">
    Confirm the correct agent answers, the caller hears audio, and Call History stores the call.
  </Step>

  <Step title="Place an outbound test">
    Confirm caller ID, audio quality, transcript quality, status, duration, and end reason.
  </Step>
</Steps>

## SIP Audio Review

SIP-routed calls use phone-grade audio. The exact quality depends on your carrier and trunk configuration.

| Evidence        | What to inspect                                                    |
| --------------- | ------------------------------------------------------------------ |
| Recording       | Noise, clipping, silence, echo, one-way audio, or bridge issues.   |
| Transcript      | Whether Deepgram or Soniox heard the caller correctly.             |
| Call status     | Whether the call connected, failed, ended early, or hit voicemail. |
| Transfer detail | Whether human handoff behaves differently through the trunk.       |
| Audio Cache tab | Whether repeated speech was served quickly or missed cache.        |

## Troubleshooting

<AccordionGroup>
  <Accordion title="Calls fail immediately">
    Check phone number format, termination URI, credentials, and whether the linked number is active.
  </Accordion>

  <Accordion title="Audio is one-way or distorted">
    Review trunk media settings, recording, and transcript. Compare against a Plivo or web call if you need a baseline.
  </Accordion>

  <Accordion title="Wrong agent answers">
    Phone number assignment controls the agent. Check inbound and outbound published versions.
  </Accordion>

  <Accordion title="Transfers behave differently">
    Test transfer behavior through the exact trunk before production use.
  </Accordion>
</AccordionGroup>

## Related Reading

<CardGroup cols={2}>
  <Card title="Phone Numbers" icon="phone" href="/calls/phone-numbers">
    Manage number status and assignments.
  </Card>

  <Card title="Audio Quality" icon="waves" href="/voice-ai/background-noise-denoising-and-audio-quality">
    Review recording and transcript quality.
  </Card>

  <Card title="Call Transfer" icon="phone-forwarded" href="/calls/call-transfer">
    Test handoffs through the trunk.
  </Card>

  <Card title="Call History" icon="activity" href="/monitoring/call-history">
    Audit SIP-routed calls.
  </Card>
</CardGroup>