A SIP server is a core component of any VoIP deployment, providing the necessary routing and control for calls to be completed. While there are many different types of SIP servers available on the market, they all operate on the same basic principles. In this article, we'll provide an overview of how SIP servers work and how you can go about setting up your own.
SIP is the primary protocol used for VoIP call setup and teardown. When a user wants to place a call, their phone will send a SIP INVITE request to the server. The server will then route the call through the network to the destination phone. Once the called party answers, media streams are established between both phones and the conversation can take place. When one or both parties hang up, another SIP request is sent to tear down the connection.
In order for all of this to work, each device needs to know where to find the SIP server. This can be done in one of two ways - either by hardcoding the address into each device (known as static provisioning), or by using a central DHCP server that hands out IP addresses and other configuration information (dynamic provisioning). Static provisioning is simpler to deploy but more difficult to manage, while dynamic provisioning offers greater flexibility but requires more infrastructure.
Once you've decided how you're going to provision devices, you'll need to choose which type of SIP server you want to use. There are three main options: open source software such as Asterisk; commercial off-the-shelf (COTS) solutions from vendors such as Cisco or Avaya; or hosted VoIP services that provide everything as a managed service. Each option has its own advantages and disadvantages - Asterisk is free but requires more technical expertise to set up and maintain; COTS solutions are expensive but offer turnkey operation; hosted VoIP services are typically lower cost than COTS but may not offer as much flexibility in terms of features and scalability.
No matter which option you choose, there are certain basics that all SIP servers must provide in order to function properly. These include support for UDP/TCP transport protocols, DNS resolution, NAT traversal, authentication and authorization, call routing rulesets, codec negotiation, session timers ,and so on . Some vendors may also offer additional proprietary features on top of these basics - for example ,Cisco's Unified Communications Manager includes built