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CT Cloud - Connectivity Requirements

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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:  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) 

  • 205.196.10.176 (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