Mobius SIP Servlet Glossary

SIP Servlet
A Java-based server-side application component that processes SIP messages (such as INVITE, BYE, REGISTER). SIP Servlets follow the JSR 116 or JSR 289 specification and are deployed in a SIP Servlet container, much like HTTP servlets run in a web container.

SIP Servlet Container
A server environment that supports SIP Servlet APIs. It receives SIP messages, invokes servlets according to routing rules, and manages the lifecycle of SIP applications. Examples include Mobius, SailFin, and Restcomm.

JSR 289
The Java Specification Request that defines version 1.1 of the SIP Servlet API. It extends earlier SIP support and introduces features like application composition and asynchronous message handling.

JSR 309
A Java API specification for media server control, allowing SIP applications to interact with a media server (e.g., to play audio, collect DTMF, handle conferencing).

WAR File (Web Application Archive)
A deployment package for Java EE web applications. SIP servlets are packaged as WAR files, containing Java classes, deployment descriptors, and any resources needed.

DAR (Default Application Router)
A configuration mechanism that defines which SIP servlet handles a given request. It uses routing rules that map SIP methods to application names and behaviors.

DAR File (restcomm-dar.properties)
A text file used by the application router to determine SIP message routing. It defines rules for methods like INVITE or REGISTER and specifies the target application and priority.

Connector
A server configuration element that defines how the container listens for incoming SIP (or HTTP) traffic. It specifies protocol type (UDP, TCP, TLS, WS), port, and IP binding.

Session
In SIP, a session refers to a real-time interaction (audio/video) established between endpoints after a successful INVITE exchange.

Media Server
A component that provides audio/video processing for services like voicemail, conferencing, IVR. Accessed by SIP servlets via JSR 309.

Load Balancer (SIP LB)
Routes SIP traffic across multiple SIP servlet containers to enable horizontal scaling and failover.

WebSockets (SIP over WS)
An alternative transport that allows SIP signaling to pass through firewalls and browsers. Often used in WebRTC setups to connect SIP to JavaScript-based clients.

WebRTC (Web Real-Time Communication)
A browser-based technology that enables peer-to-peer audio, video, and data communication without plugins. Often paired with SIP Servlets for signaling.

SIP Transaction
The request-response exchange between two SIP endpoints. A transaction is uniquely identified by its branch ID and method.

Dialog
A persistent relationship between two SIP endpoints, established through INVITE and maintained with responses. It allows tracking and correlation of all messages in a session.

User Agent (UA)
A SIP endpoint that sends and receives SIP messages. A softphone, IP phone, or browser

Softphone
A software-based SIP user agent used for testing or actual communication.