Getting Started with Telnyx

Telnyx SIP proxies are available in the following regions:

SIP Regions affect only the signaling path. To control where media is anchored, see AnchorSite™ below.

You can configure your SIP server or device to send calls to any SIP region listed above. Use the FQDN or IP addresses depending on your device's requirements.

The FQDNs for each region support DNS record types A, SRV, and NAPTR. DNS A records resolve to IP address 1. SRV and NAPTR records provide round-robin routing across both IPs and advertise available transport protocols and ports.

⚠️ Outbound calls from the same IP address or SIP username are limited to 20 calls per second. Telnyx reserves the right to adjust this limit.

Telnyx delivers inbound calls to your SIP infrastructure using one of two methods:

If your device uses SIP registration, register to any SIP region listed above.

For IP or FQDN authentication, select your inbound SIP region in the Inbound section of your Connection in Mission Control. If no SIP Region is explicitly configured, Telnyx selects it based on your AnchorSite™ setting. This determines which proxy IPs deliver calls to your infrastructure.

⚠️ If you use an ACL or firewall, allow-list all IP addresses for your selected SIP Region.

Telnyx supports the following transport protocols for SIP signaling:

The signaling addresses, transport protocols, and media CIDRs listed on this page are also available as a machine-readable JSON file:

https://sip.telnyx.com/voice.json

Use this endpoint to programmatically keep your firewall rules, SIP configuration, or monitoring tools in sync with the latest Telnyx network information.

Telnyx proactively pings your connection using ICMP ping messages to measure round-trip times from all available sites. The PoP with the lowest latency is automatically selected.

⚠️ Allow the Telnyx media IP addresses listed in this article on your firewall so ICMP pings can reach your endpoint.

If you know which PoP best serves your setup, you can anchor your media to a specific site. This is recommended when your media IP is in a different region than your signaling IP (latency detection is based on signaling IP) or when ICMP is not supported.

The RTP port range used by Telnyx is 16384 to 32768 (UDP).

Telnyx uses the following media IP addresses for RTP streams.

⚠️ If you use an ACL or firewall, allow-list these addresses:

36.255.198.128/25
50.114.136.128/25
50.114.144.0/21
64.16.226.0/24
64.16.227.0/24
64.16.228.0/24
64.16.229.0/24
64.16.230.0/24
64.16.248.0/24
64.16.249.0/24
103.115.244.128/25
185.246.41.128/25

Telnyx supports the following audio and video codecs:

• G.711U (PCMU)
• G.711A (PCMA)
• G.729
• G.722 (wideband)
• Opus (wideband)
• AMR‑WB (wideband, HD Voice)
• H.264 (video)
• VP8 (video)

Telnyx uses a 20 ms packetization interval (ptime=20) for RTP media.

1. Log in to Telnyx Mission Control.
2. Create a Connection for your SIP device. Choose from the following authentication types:
   • Credentials (username and password) — inbound and outbound
   • IP address — inbound and outbound
   • FQDN (inbound) + Credentials (outbound)
   • FQDN (inbound) + IP address (outbound)

1. Create an Outbound Profile and assign your Connection to it.
2. Configure your SIP device to send calls to sip:sip.telnyx.com:5060 (UDP or TCP) or sip:sip.telnyx.com:5061 (TLS)
3. Destination numbers in +E.164 format are supported by default. You can configure regional settings on your Connection to allow local dialing formats.

1. Purchase a number and assign it to your Connection.
2. If you chose Credentials authentication, register your SIP device with your username and password to sip:sip.telnyx.com:5060 (UDP or TCP) or sip:sip.telnyx.com:5061 (TLS) before receiving calls.
3. With IP Authentication, Telnyx delivers calls from one of your inbound SIP Region's IP addresses (see table above) using UDP by default. To use TCP or TLS, edit the SIP Transport Protocol option for Inbound Calls on the Connections page.
4. You can enter multiple IP addresses with different priorities. Telnyx attempts each address in priority order, respecting the No Ringback Timeout setting.
5. With FQDN Connections, Telnyx routes calls to the resolved FQDN records (SRV records supported).

Telnyx supports fax over G.711 or T.38.

For inbound fax, Telnyx expects a T.38 re-INVITE from the customer by default. If no re-INVITE is received, the call continues with G.711. Disable this via the T.38 option on the Numbers page.

For outbound fax, choose a fax mode using the T.38 Re-invite Initiated By options in the Outbound tab on the Connections page:

• Telnyx Telnyx sends a T.38 re-INVITE after detecting a fax tone
• Customer Telnyx expects a T.38 re-INVITE from the customer
• DisabledTelnyx won't send a T.38 re-INVITE and will reject any received

Telnyx provides STUN service using the following addresses:

stun.telnyx.com:3478

Telnyx provides TURN service using the following address:

turn.telnyx.com:3478

Request TURN credentials by emailing support@telnyx.com.

SIP over WebSockets enables real-time voice calls directly from the browser, with no plugins required. The browser handles audio and media natively, while SIP signaling is carried over a WebSocket connection.

Use a third-party SIP JavaScript library (such as SIP.js or JsSIP) to connect directly to Telnyx's SIP WebSocket endpoints, using the same SIP regions listed above.

Example for US SIP region:

wss://sip.telnyx.com:7443 (encrypted)