If you have a SIP enabled PBX, you may not realize just how many components work in conjunction to provide you with quality service. When you’re placing calls, one of the most important aspects to make everything run smoothly is the SIP proxy server. It is what handles beginning and terminating your SIP call.
First of all, what is SIP Trunking? SIP, which stands for Session Initiation Protocol, combines your existing voice and data networks into a unified network. Not only does this offer great communications, but it typically comes with significant cost savings. You can even use SIP with your current PBX systems.
SIP Trunking comes with many features not found on all Business Phone Systems. For example, with SIP you can enjoy caller ID, Direct Inward Dial (DID), cell phone integration, find me/follow me, voicemail transcription, and audio/video conference bridges. By taking advantage of the aforementioned features, your business can prove more efficient and productive.
SIP also uses endpoints (also known as user agents) that can begin, modify, or terminate a session. Your endpoints can even be softphones, cell phones, or computers.
SIP Proxy Server
A SIP Proxy Server is what handles the management of SIP calls within your network. It takes requests from your user agents to make and end calls. If you have a SIP enabled IP-PBX, it usually comes with a SIP Server.
How Does it Work?
The best way to think of a SIP Proxy Server is that it is the middleman of your SIP calls. It makes and ends calls via stateless and stateful servers.
Stateless servers do not retain information. As soon as it forwards a message it receives, it ‘forgets’ that the request ever happened.
On the other hand, a stateful server stores information. It keeps track of requests that have been received. Stateful servers can store them for future use.