Intro

This articles explains what port ranges will need to be used, open, and configured with WIN-911 S/I/A when working with the specific VoIP providers and SIP providers that WIN-911 support.


PLEASE NOTE - You will need to engage your IT department. Configuring a VoIP system will require a coordinated effort between your company's voice network experts (often IT or network engineers) and the party responsible for installing WIN-911. It is STRONGLY RECOMMENDED to engage your VoIP network experts early in the process.



Port ranges for Skype connect: https://support.skype.com/en/faq/FA148/which-ports-need-to-be-open-to-use-skype-on-desktop

443/TCP

3478-3481/UDP

50000-60000/UDP

1000-10000/TCP

50000-65000/TCP

16000-26000/TCP



Port ranges for voiptalk: https://www.voiptalk.org/products/configuration-of-trixbox-behind-a-nat-firewall-setup.html

UDP Port 5060 is for SIP communication

UDP Port 5060-5082 range, SIP communications

UDP Port 10000 - 20000 is for RTP - the media stream, voice/video channel



Port ranges for AxVoice: https://www.axvoice.com/support/faq/faq.html - Internet & Local Network > What ports do I need to ensure are open on my firewall?

The VoIP phone service uses several OUTBOUND UDP connections utilizing Ports 5004-5065 If you have questions regarding the compatibility of your router with your new VoIP service you should contact the router manufacturer directly or the Broadband Internet Service Provider Company.



Port ranges for surevoip: http://www.surevoip.co.uk/support/wiki/nat_and_firewall_settings

For Deskphones, allow ports 5060 UDP and 10000 to 40000 UDP to pass through your firewall to access your phones



Port for Gafachi: http://www.multitech.com/documents/publications/stories/gafachi.pdf

UDP Port 5060



Port ranges for Ozeki Phone System XE: http://www.ozekiphone.com/voip-how-to-setup-your-pbx-to-go-through-firewall-77.html

UDP Port 5060

RTP Port 5000 - 10000 range



Port ranges for Trixbox: https://www.voiptalk.org/products/trixbox-behind-nat-firewall-setup

UDP Port 5060 is for SIP communication.

UDP Port 5060-5082 range, SIP communications.

TCP Port 5060 is for SIP but thought to be rarely used.

UDP Port 10000 - 20000 is for RTP - the media stream, voice/video channel.



Port ranges for Cisco Unified CM (Call Manager): https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/port/9_0_1/CUCM_BK_T98E8963_00_tcp-port-usage-guide-90/CUCM_BK_T98E8963_00_tcp-port-usage-guide-90_chapter_01.html



Port ranges for OpenSER (Kamailio): https://kamailio.org/dokuwiki/doku.php/core-cookbook:1.3.x

Source and destination port are 5061

Advertised port is 5080

TCP port range is 5060 - 5064

0 means that it will connect to any port

Rewrite port number is 5070



Port ranges for Cisco CM Express: https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unity-express/62609-tdcmecue.html#t6

Default port range for IP phone registration is 2000



Port ranges for PBXnSIP: https://manualzz.com/doc/7292446/pbxnsip-admin-manual

SIP port ranges are 5060 - 5062

PTSN port range is 2048 - 2096

Binding port is 8080

RTP port ranges are 49152 - 64512

SNMP default port is 161

TFTP default port is 69



Port ranges for Asterisk: https://wiki.freepbx.org/pages/viewpage.action?pageId=24051965#SIPAudioIssuesFreePBX12+-OpentheSIPandRTPportstoyourAsteriskserver

For SIP protocol, open UDP (NOT TCP) port 5060 (SIP)

Open ports 10000-20000

Open UDP port 4569 (IAX)


Port ranges for sipXecs: https://sipfoundry.atlassian.net/wiki/spaces/sipXecs/pages/492152/Quick+Start

SRV records for the SIP communications (port 5060 tcp & udp)

SRV record for the resource record (port 5070 tcp)

SRV record for XMPP client connections (port 5222 tcp)

SRV record for XMPP server connections (port 5269 tcp)

SRV record for XMPP client connections to XMPP conference (port 5222)

SRV record for XMPP servers connections to XMPP conference (port 5222)

Please follow guidance in link below:

http://wiki.sipfoundry.org/display/sipXecs/DNS+Concepts+for+sipXecs


Port ranges for PBXpress: http://docs.globalink.us/pbxpress/PBXpress_Admin_Guide.pdf

5050 for SIP


Port ranges for FreeSwitch: https://freeswitch.org/confluence/display/FREESWITCH/Firewall#Firewall-TypicalPorts

1719    UDP

1720    TCP

3478    UDP

3479    UDP

5002    TCP

5003    UDP

5060    UDP & TCP    SIP UAS    Used for SIP signaling (Standard SIP Port, for default Internal Profile)

5070    UDP & TCP    SIP UAS    Used for SIP signaling (For default "NAT" Profile)

5080    UDP & TCP    SIP UAS    Used for SIP signaling (For default "External" Profile)

8021    TCP     ESL    Used for mod_event_socket *

16384-32768    UDP


Port ranges for FreePBX: https://wiki.freepbx.org/pages/viewpage.action?pageId=24051965#SIPAudioIssuesFreePBX12+-RTPPortRange

For SIP protocol, open UDP (NOT TCP) port 5060 (SIP)

Open ports 10000-20000 (RTP)

Open UDP port 4569 (IAX)


Port ranges for SwyxWare: https://www.swyx.com/products/support/knowledge-base/article-details/swyxknowledge/kb2308.html

SwyxIt!                            50000-50099

SwyxPhone L400             5010/11

SwyxPhone L420             5006/7

SwyxPhone L420e,s        5010/11

SwyxPhone L440            5010/11

SwyxPhone L100            4103-4119

SwyxServer                    51000-51499

SwyxGate                      51500-51999

LinkManager                   55000-56000

ConferenceManager        56000-57000

PhoneManager               no data ports

ITSPManager                 no data ports

SwyxFax Server             2000-8000,61000


Port ranges for Aastra MX-One: http://www.tango-networks.com/wp/partner/?wpdmact=process&did=%2F7.0+PBX+Integration+Guides%2FAll%2FAccelerator+Aastra+Integration+Guide.pdf

Aastra MX-One PBX with a port value of 5060 SIP



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