---
title: "CT Cloud - Connectivity Requirements"
slug: "ct-cloud-connectivity-requirements"
tags: ["calltower", "public"]
updated: 2026-09-18T14:53:20Z
published: 2026-09-18T14:53:20Z
canonical: "learn.calltower.com/ct-cloud-connectivity-requirements"
---

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

# CT Cloud - Connectivity Requirements

## Overview:

In general, very little setup is required to get a site ready for CT Cloud Voice, SIP, and Communicator. The following information can be used for more restrictive environments or for troubleshooting purposes.

## Network (CT Cloud Voice and Communicator)

> [!NOTE]
> Note:
> 
> **IF YOUR NETWORK ROUTER OR FIREWALL HAS A SIP APPLICATION LAYER GATEWAY (SIP ALG) YOU MUST DISABLE IT. REFER TO YOUR FIREWALL OR ROUTER VENDOR FOR MORE INFORMATION.**

SIP protocol for Desktop phone: UDP SIP signaling for desktop phone: 15060 DTMF: RFC2833 Ports for RTP (audio): UDP 16384 thru 65535

SIP protocol for Accession desktop: UDP SIP signaling for Accession desktop: 5100 DTMF: RFC2833 Ports for RTP (audio): UDP 16384 thru 65535

SIP protocol for Accession mobile: TCP SIP signaling for Accession mobile: 443 DTMF: RFC2833 Ports for RTP (audio): TCP 16384 thru 65535

**FIREWALL ADJUSTMENTS FOR ALLOWED IP ADDRESSES (if necessary for your infrastructure):**

- 205.196.170.145 (signaling and RTP)
- 205.196.171.145 (signaling and RTP)
- 205.196.170.163 (SSL for phone provisioning)
- 205.196.170.176 (SSL for phone provisioning)
- 205.196.171.183 (IM/SMS if using chat or SMS via Accession)
- 205.196.171.184 (IM/SMS if using chat or SMS via Accession)

## SIP

1. **Digest**
  1. **Primary:** 205.196.170.135
  2. **Secondary:** 205.196.171.135
  3. **Protocol:** UDP (default) or TCP
  4. **Signaling port:** 5060
  5. **RTP ports:** UDP/TCP 16384 thru 65535
2. **IP Auth**
  1. **Primary:** 205.196.170.136
  2. **Secondary:** 205.196.171.136
  3. **Protocol:** UDP (default) or TCP
  4. **Signaling port:** 5060
  5. **RTP ports:** UDP/TCP 16384 thru 65535

## Provisioning:

**Although desk phones should provision automatically, you may want to configure DHCP option 66 in your network DHCP server. Option 66 should be set to** [**https://commportal.calltower.com/sip-ps**](https://commportal.calltower.com/sip-ps)

A telephony service that operates over the internet, allowing users to make and receive calls using cloud technology.

Session Initiation Protocol, is the technology that starts, manages, and ends internet-based phone calls and video meetings between people and devices.

A specific service offering from CallTower that provides cloud-based voice communication solutions.