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SIP is the abbreviation of 'Session Initiation Protocol'. In simple terms it is a standards based protocol that allows voice and video communication over the internet.
SIP cannot make a VoIP calls on its own. It just takes care of introducing and closing the phone call. It does this in two stages:
SIP Registration to the VoIP provider
Establishes connection and confirms you are ready to make/receive calls.
SIP Invite conversation
When you actually make/receive calls from your SIP phone
The first step in making VoIP call using SIP is SIP registration. This takes place when your softphone or hardware phone registers with your VoIP provider. It’s like saying:
“Hi, I’m here and ready to make and receive phone calls”.
Most softphones and hardware phones register with a session time usually set between 1 and 60 minutes. When that period finishes, it will register again. It’s VoIP’s way of saying:
“Hey I’m still here, just wanted to check in with you again. I’m still here!”.
There is no loss in communication during this time and does not affect your calls.
A SIP Invite Conversation takes place when you actually make and/or receive a call. The most important part of this is “Invite”.
The “Invite” is where your client, and your VoIP Provider’s servers, are negotiating the status of the call. The “Invite” is establishing what codecs both parties are using and in what order of preference to access them. It also agrees which RTP (Real-time Transport Protocol) ports are going to be used to transmit audio to and from each other.
SIP ALG (Application Layer Gateway) is a feature on most routers that is intended to assist users on private IP addresses, but in many cases it is implemented poorly and can actually cause more problems than it solves!
When SIP ALG is active it can modify the SIP packets of calls in unexpected ways, corrupting them and making them unreadable.
This video explains the symptoms you might experience with SIP ALG and how to disable it.
Traditional phone calls are made over the PSTN 'Public Switched Telephone Network‘ which is a network of physical copper phone lines that need to be connected to allow telephones to make calls.
SIP Trunks are virtual phone lines that allow you to make and receive VoIP Calls over the internet to anyone around the world. It doesn’t matter whether they are using VoIP or traditional PSTN, as long as they have a phone number you can call them.
SIP Trunks use a packet switch network that converts your voice into digital packets and sends it across the network to the recipient.
Experience SIP for yourself with a free business phone system trial today.