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)
205.196.170.176 (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
Digest
Primary: 205.196.170.135
Secondary: 205.196.171.135
Protocol: UDP (default) or TCP
Signaling port: 5060
RTP ports: UDP/TCP 16384 thru 65535
IP Auth
Primary: 205.196.170.136
Secondary: 205.196.171.136
Protocol: UDP (default) or TCP
Signaling port: 5060
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