SIP stands for Session Initiation Protocol. It is used to initiate, maintain, and terminate real-time communication sessions that include voice, video, and instant messaging. Applications of the SIP protocol include signaling and controlling multimedia communication sessions such as VoIP calls, video calls, and mobile phone calls over LTE. A SIP server is the main component of an IP PBX and manages all SIP calls in the network. A SIP server is also called a SIP proxy, SIP proxy server, or registrar.

A SIP proxy sits between endpoints. It receives a request, checks rules, applies security, and sends the request to the correct destination. That destination can be a phone on the same network, a phone in a remote office, or the public telephone network.

A SIP server manages sessions. When an IP phone places a call, the phone sends an INVITE request to the proxy server. The proxy server reads the header, confirms the caller, locates the callee, and forwards the request. It sets up a session between two or more endpoints, negotiates the media parameters for each endpoint using SDP, and can adjust those parameters while the call is in progress. It can also substitute one endpoint for another and end the session when either side hangs up.

The proxy server applies call rules. You can let multiple phones ring at the same time, ring them in order, or send the call to voicemail after a set time. Because the rules live in the proxy, you can change them without touching every phone.

During a session the proxy server can receive a request to place the call on hold, transfer it, or park it. The proxy reads the new request, checks policy, and forwards it. This flexibility shows why the proxy is often called the control brain of a VoIP network.

The SIP proxy does not carry audio or video. Media travels by RTP between endpoints or through a dedicated media server. In most small or medium deployments the SIP proxy and media server run on the same machine to simplify management. In high-volume services the operator may run many media servers behind a front-end SIP proxy to share load.

Keeping media outside the proxy removes heavy packets from the control path. That design lets one SIP proxy server handle thousands of calls while a cluster of media servers moves the voice and video streams.

A proxy server blocks unauthorized traffic before it reaches phones. It checks every REGISTER or INVITE for valid credentials. When a user calls an international number or any charged service, the proxy server can check credit and mark the call for billing. These checks prevent fraud and warn the provider about suspicious patterns.

Transport Layer Security can protect signaling while Secure RTP can protect media. Many operators also deploy SIP Outbound or keep-alive pings so the proxy knows when a phone drops offline and can route new calls to voicemail.

A stateless SIP proxy server forwards each request and forgets it. The approach is fast and light on memory. A stateful SIP proxy server keeps dialog data. It can fork one call to several endpoints, record call history, and handle complex failover. Large carriers often mix modes: a public-facing stateless proxy for speed and an internal stateful proxy for rich features.

In a company network the SIP proxy server links desk phones, mobile softphones, WebRTC browsers, and PSTN trunks. It provides the same dialing plan to every site, keeps presence status, and routes instant-messaging messages. Developers can call a REST or WebSocket API on the server to pop a CRM record or start a recorded call.

Many businesses replace legacy trunks with SIP trunking. A SIP trunk is an IP link that carries many concurrent calls. The SIP proxy server packs calls into the trunk and cuts rental fees. Because the same proxy server handles local and trunk traffic, you get one set of rules for internal and external calls.

Running your own proxy server gives control, flexibility, and cost savings.

• Central call control keeps features consistent across offices.
• Scaling is simple: add phones and update the dial plan in one place.
• SIP trunking reduces monthly spend compared with legacy lines.
• Logs help spot misuse fast and aid compliance checks.
• Standard protocols avoid vendor lock-in and work with many phones.

When planning a VoIP service place the proxy on a stable public IP or behind a fire-wall with correct port forwarding. Keep the media server close to the proxy to avoid delay. Use DNS-SRV records so phones can find the proxy even if the IP changes.

For high availability run two SIP proxy servers. A heartbeat tool watches the active node and moves traffic to the standby node if the active node fails. Use a shared database for registrations so phones do not need to re-register after failover.

You can design many rules. An office reception phone can ring first, then a hunt group, then a mobile. A help line can fork to every support agent at once. A night rule can send calls to voicemail outside hours. All logic lives in the SIP proxy server so changes need no firmware update.

During call setup each endpoint lists the codecs it supports. The proxy server forwards the lists and the endpoints pick a match such as G.711, Opus, or G.722. If one endpoint pauses the call, the proxy server updates the session to remove media paths until the call resumes. Quality tools such as RTCP reports help the proxy spot bad links and switch calls to a backup trunk.

WebRTC clients wrap SIP messages in WebSocket frames. The SIP proxy server unwraps them and bridges to classic SIP endpoints. Browsers then talk to desk phones or mobile apps without plug-ins. Because media stays in RTP the media server needs no change.

A single code base can host many tenants. Each tenant has a domain, dial plan, voicemail store, and separate trunks. The proxy server keeps the data in a shared database and applies it when a request arrives. Service providers use this model to supply hosted PBX services to many small businesses.

SIP MESSAGE requests carry chat text. The SIP proxy server stores the chat in logs, tags it with time, and sends it to the receiver. Presence events such as BLF or call-state also pass through the proxy, letting phones show user status.

• Verify DNS and SRV records point to the correct proxy host name.
• Open SIP and RTP ports on the fire-wall.
• Set TLS certificates.
• Configure codecs and trunk limits.
• Test internal, outbound, and inbound calls.
• Check hold, transfer, and voicemail.
• Review call detail records for accuracy.

Most SIP servers follow open standards, which means they work with a wide range of SIP phones, softphones, and VoIP providers. This flexibility lets businesses mix and match devices, scale over time, and avoid vendor lock-in. A well-configured SIP proxy server supports both IPv4 and IPv6, handles NAT traversal, and works with TLS and SRTP for secure signaling and media. Whether used for internal calls, external communication, or hybrid setups, the SIP proxy remains a key part of any reliable VoIP deployment.

SIP server: software that controls sessions.
SIP proxy: same server viewed as a traffic director.
SIP proxy server: full term for search engines.
Proxy server: general term for any middle box that forwards requests.
Stateless proxy: forgets dialogs.
Stateful proxy: tracks dialogs.
VoIP: voice over IP.
Network: the IP path that carries signaling and media.
Benefits: gains such as cost reduction, control, and scale.