# Advanced Settings of SIP Extension

Every SIP (Session Initiation Protocol) extension exposes a group of advanced options that control how the phone system exchanges call signaling and audio with the registered device (a desk phone, a softphone, or the Cloud Voice App). You will find these options on the extension's own configuration page, grouped together under its advanced settings. Leave them at their defaults unless you have a specific device or network reason to change them, and change one setting at a time so you can tell what caused any change in behavior.

:::caution
These options assume a working knowledge of the SIP protocol. Setting them incorrectly can break calling on the affected extension.
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## DTMF Mode

DTMF (Dual-Tone Multi-Frequency) is the touch-tone signal a phone sends when a caller presses a key, for example when choosing an option in an auto attendant menu or entering a voicemail PIN. This setting controls how those key presses are carried across the call:

- **RFC4733 (RFC2833)**: The digits travel inside the RTP (Real-time Transport Protocol) media stream as their own dedicated packets, separate from the voice audio. RFC4733 is the current standard and replaces the older RFC2833.
- **Info**: The digits are sent as SIP INFO messages on the signaling channel instead of in the media stream.
- **Inband**: The digits are encoded directly into the audio, the same way a person would hear the tones.
- **Auto**: The system uses RFC4733 (RFC2833) when the device reports support for it, and falls back to Inband when it does not.

:::tip
If callers reach a menu or voicemail but their key presses are ignored or misread, the DTMF mode usually does not match what the device or the connected provider expects. RFC4733 (RFC2833) is the most reliable choice for VoIP, so try it first.
:::

## Transport

Transport is the network protocol that carries the SIP signaling messages (the call setup and teardown, not the audio itself). Select one or more of the following:

- **UDP** (User Datagram Protocol): the traditional, lightweight default for SIP.
- **TCP** (Transmission Control Protocol): a connection-based option that handles larger messages more reliably.
- **TLS** (Transport Layer Security): TCP with encryption, so the signaling cannot be read in transit.

:::note
- **TCP** can be selected only after the SIP TCP port has been turned on under **PBX Settings > SIP Settings > General > Basic**.
- **TLS** can be selected only after TLS has been turned on under **PBX Settings > SIP Settings > TLS**.
:::

When you enable more than one protocol, the system decides which to use based on the device type:

- **Cloud Voice App and auto-provisioned IP phones** follow the priority TLS > TCP > UDP.
- **Manually registered IP phones** use whichever protocol the phone specifies. If the phone does not specify one, the same default priority applies: TLS > TCP > UDP.

:::caution
The device at the other end must support the transport you select, or it will fail to register and the extension will not be able to make or receive calls. Confirm the phone or app is configured for a matching protocol before you remove UDP or otherwise narrow the list.
:::

## Qualify

Turn this on to have the system periodically send a SIP OPTIONS packet (a lightweight "are you still there?" request) to the device and wait for a reply.

:::note
This lets the system track whether the extension is reachable and show it as online or offline. On networks that use NAT (Network Address Translation), the regular OPTIONS traffic also helps hold the connection open through the router so incoming calls keep reaching the device.
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## T.38 Support

T.38 is a protocol for sending faxes reliably over an IP network by carrying the fax signals as data rather than as audio. This setting turns T.38 handling on or off for the extension.

:::tip
T.38 adds processing overhead, so leave it disabled unless the extension actually needs to send or receive fax over T.38. Disabled is the recommended setting.
:::

## SRTP

SRTP (Secure Real-time Transport Protocol) encrypts the call audio so it cannot be intercepted and listened to on the network. Turn it on to protect the media on this extension.

:::tip
SRTP only encrypts the audio. For fuller protection, pair it with the TLS transport option above, which also encrypts the signaling. Both the extension and the device must support SRTP for an encrypted call to connect.
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