1c:I[12781,["2094","static/chunks/f41695f6-dee59cfd36c73118.js","5692","static/chunks/8519c14d-8c52348b3557d852.js","4274","static/chunks/4274-56609c54fe70d805.js","8966","static/chunks/8966-36c202f453ff6c2e.js","2325","static/chunks/2325-c32c1073d914498e.js","2781","static/chunks/2781-cc683d9d661721e3.js","1385","static/chunks/app/(site)/(layout)/(standard)/faq/%5Bcategory%5D/%5Bquestion%5D/page-fb4a7e399d22dd88.js"],"default"] 1d:T767,

VoIP calls consist of three major steps:

1) A SIP registration to the Yay Platform so we know you are connected to us and ready to make and receive calls

2) A SIP Invite Conversation when you actually make and or receive a call

3) The actual audio of the call itself which takes place within the “SIP Invite" conversation but using RTP to communicate the audio to and from you and Yay.

Steps 1 & 2 take place over UDP or TCP Port 5060 if un-encrypted or 5061 over TCP, if encrypted. Encryption is also known as TLS and you need to activate encryption on both your phone and in the Yay Dashboard at Dashboard > My VoIP > SIP Users if its being used.

Step 3 can pretty much use any port you wish. The Yay platform will always use a port between 10,000 and 40,000 and your softphone or hardware phone will define which port it wishes to use.

In all cases, the above is detail you shouldn’t really need to know and that’s because each part of VoIP call is initiated by each side. So nothing special needs to be done to your firewall to make a call happen.