Session Initiation Protocol (SIP) is a signaling protocol used to initiate, maintain and terminate multimedia communication sessions such as voice and video calls over the Internet. It is similar to HTTP but for real-time communications, serving as the backbone for most VoIP (Voice over IP) services. SIP manages these sessions but doesn't handle the media itself. Components of SIP include User Agent Clients (UAC) and User Agent Servers (UAS), which communicate through requests and responses to set up, modify and end sessions. This protocol is widely adopted for its scalability and integration with other protocols, supporting a broad range of communication services.
SIP (Session Initiation Protocol) is a signaling protocol used to control multimedia communication sessions, such as voice and video calls, over Internet Protocol (IP). SIP is analogous to HTTP for voice and is essentially the glue that ties communications systems together, much like HTTP ties clients and servers together for worldwide communication. More and more vendors are implementing SIP as a standard telephony platform. This appendix provides an introduction to SIP and is designed to introduce you to the key concepts and mechanisms of the SIP protocol
What is SIP?
SIP was originally designed in 1996 to create a mechanism for inviting people to large-scale multipoint conferences on the Internet multicast backbone. The first version (SIP 2.0) was defined by RFC 2543, and by November of 2000 the protocol had been refined and clarified in RFC 3261. Though IP telephony didn't actually exist at the inception of RFC 3261, SIP evolved to provide a missing piece of Internet architecture: a way for users to explicitly invite others users to join sessions over the Internet.
What is a Session initiation protocol?
SIP is a signaling protocol used to control multimedia communication sessions, such as voice and video calls, over Internet Protocol (IP). SIP is analogous to HTTP for voice and is essentially the glue that ties communications systems together, much like HTTP ties clients and servers together for worldwide communication. More and more vendors are implementing SIP as a standard telephony platform as its popularity increases.
What is a SIP Session?
A SIP session is a related progression of events devoted to a particular activity occurring over the Internet. Activities can include two-way telephone calls, video conferencing, streaming multimedia distribution, instant messaging, presence and online games.
SIP knows nothing about the details of the sessions it controls: it only initiates, terminates and modifies the sessions. Although SIP can work in a framework with other protocols — SOAP, HTTP, XML, VXML, WSDL, UDDI and SDP — it does not perform any of their functions.
SIP Components
User Agent Client (UAC): The UAC generates "methods" and sends them to servers (e.g., it sends an INVITE request and initiates a call).
User Agent Server (UAS): The UAS receives the methods, processes them, and generates responses(e.g., it sends a 200 Ok response to indicate a successful session).
The UAS may issue multiple responses to the UAC.
User Agent Clients
User Agent Clients: The UAC is often associated with the end-user, since applications running on systems are used by people. The UAC can be any end-user device, such as a cell phone, multimedia handset, personal computer (PC), personal digital assistant (PDA) or a softphone. The requests generated by the UAC are sent to a server (typically a proxy server) and are known as "meth-ods," which will be discussed later.
Note: Non-IP devices like dumbphones can also be turned into SIP UAs by using an inexpensive analog telephone adapter (ATA) to make them SIP-aware. An ATA is a box with one or two analog ports with RJ11jacks used to connect regular analog phones to the VoIP network. Popular ATAs include the SIPura or Linksys SPA 112.
User Agent Servers
Servers possess a predefined set of rules to handle the requests sent by clients and are usually part of the network. There are several types of servers:
Proxy Server — Proxy servers help track down addresses of recipients whose exact addresses aren't known in advance. If the proxy server cannot find the address of the recipient, it will send the request to other proxy servers. Destinations include another extension on the same proxy server, the next-hop proxy server in the routing table or a media server. SIP proxy servers use presence services to track users, which means users can be located regardless of physical location. Proxy servers are the most common server in the SIP environment.
Registrar Server — A SIP registration server is responsible for registering devices. It does this by authenticating the device with a user name and password and keeping a table of IP addresses and extensions/phone numbers. This authentication process is similar to logging in to a web server, which requires a user name and password. The registrations server makes it possible for users to alter the address at which they can be contacted. Registrations play an important role in the process, since SIP devices that do not register cannot be called, and SIP devices that do not successfully authenticate cannot make outbound calls. A media server is a device that handles any kind of media or RT, such as a voicemail server, a conference server, an IVR server or a music on hold server.
Following a critical PHP security advisory affecting installations worldwide, we temporarily restricted public access to some of our backend services while carrying out necessary upgrades. Access has now been restored, and the affected services are operating normally. The Vodia PBX itself does not use PHP and was not directly affected. We explain what happened, the changes to legacy licensing and TLS support, and why customers running older PBX versions should contact us for upgrade guidance.
Submitting a support ticket with the right information helps our team understand your issue and reduces follow-up questions. This step-by-step guide walks you through the Vodia Help Center, from describing the problem and attaching diagnostic files to setting the business impact and providing your PBX details. You’ll also learn which call details and timestamps to include, alongside practical tips to keep troubleshooting focused and help our support team investigate your issue more efficiently.
Switching phone systems doesn’t have to mean rebuilding everything from scratch. The Vodia PBX migration tool imports key configuration data from backups, including extensions, ring groups, queues, desk phones, and basic auto-attendant settings. Built into the Vodia Admin Portal, it guides administrators through the import process and reduces repetitive manual work, whether you’re moving 10 extensions or 1,000. Explore what transfers, what needs manual setup, and how the built-in tool reduces repetitive work when moving to Vodia.
.svg)
