SIP — Session Initiation Protocol

Session Initiation Protocol (IETF RFC 3261)

For DSP telephony integration, see audio/dsp-fundamentals. For paging system design including SIP paging, see audio/paging-systems.

SIP (Session Initiation Protocol) is the IETF-standard signaling protocol for establishing, managing, and terminating voice and video calls over IP networks. It is the protocol underlying nearly all VoIP phone systems, SIP trunks to telephone carriers, SIP-based paging systems, and many video conferencing endpoints. In AV systems, SIP appears most commonly in DSP telephone interfaces (Biamp Tesira, Q-SYS soft phone), SIP paging controllers, and SIP-based video endpoints (Poly, Cisco, Yealink). AV integrators do not need to be telephony engineers, but understanding SIP basics helps with configuration, troubleshooting, and coordinating with IT and telecom teams.

How SIP Works

SIP is a text-based protocol similar to HTTP. It handles only call signaling — setting up and tearing down calls, negotiating media formats, and routing calls through the network. The actual audio or video is carried separately by RTP (Real-time Transport Protocol). See glossary/rtp-rtsp.

Key SIP components:

• SIP User Agent (UA): the endpoint — a SIP phone, DSP telephone interface, or softphone
• SIP Registrar/Proxy: the server that handles registration, routing, and call setup (often the PBX or cloud UC platform)
• SIP Trunk: the connection from an on-premise PBX to a PSTN (phone network) carrier via SIP instead of traditional phone lines

Typical call flow:

1. UA registers with the SIP registrar (sends a REGISTER message with credentials)
2. Caller sends INVITE to the registrar
3. Registrar routes INVITE to the callee's UA
4. Callee responds with 200 OK; media (RTP) begins flowing directly between UAs
5. Either party sends BYE to terminate the call

SIP in DSP Telephone Interfaces

DSPs like Biamp Tesira (with VoIP card), Q-SYS (Softphone component), and Extron DMP (with optional VoIP card) can register as SIP endpoints on the organization's PBX or SIP trunk. This allows the DSP to:

• Receive inbound calls and route audio to the room
• Dial out from the room to any phone number
• Integrate with conference bridge dial-in numbers

Configuration requires: SIP server address (PBX IP or SIP proxy), SIP username/extension, password, and SIP domain. SIP authentication failures (401 Unauthorized) indicate incorrect credentials. SIP registration timeouts indicate network routing problems (often a firewall blocking SIP ports UDP 5060 / TCP 5061).

SIP in Paging Systems

SIP paging controllers (Algo, CyberData, Barix) register as SIP extensions on the PBX. When the PBX dials the paging extension, the controller triggers the paging amplifier and loudspeakers. SIP paging enables:

• Zone paging triggered by dialing a paging extension from any phone
• Integration with emergency notification systems that can dial SIP endpoints
• Multi-zone paging by dialing multiple SIP extension groups simultaneously

Common Pitfalls

• SIP registration failing due to firewall blocking. UDP port 5060 (SIP signaling) and the RTP port range (typically UDP 10000–20000) must be open between the DSP/endpoint and the SIP proxy. A common IT error is opening SIP signaling but not the RTP range, resulting in one-way audio. Fix: verify both SIP (5060/5061) and RTP port ranges are permitted; check with IT for any NAT traversal requirements (STUN/TURN may be needed for cloud-hosted PBX).

• One-way audio on SIP calls. The call connects but audio flows only in one direction. Typically an RTP port range firewall issue or a NAT traversal problem (the DSP is behind NAT and sends the wrong IP in the SDP). Fix: verify RTP ports are open; if behind NAT, configure STUN server in the SIP endpoint settings or enable symmetric RTP.

• SIP endpoint re-registers every few minutes, causing call drops. The SIP registration expiry time is too short, or the PBX is not refreshing registrations. Some firewalls also break SIP registration keep-alives. Fix: increase SIP registration expiry (REGISTER Expires header) to 3600 seconds; configure the PBX to allow the endpoint's registration interval; disable SIP ALG (Application Layer Gateway) on any firewall in the path — SIP ALG frequently breaks SIP signaling.

• SIP ALG corrupting SIP messages. Many consumer and small business routers/firewalls include SIP ALG as a "helpful" feature that rewrites SIP headers. In practice, SIP ALG mangles SIP messages and causes call setup failures and one-way audio. Fix: disable SIP ALG on all routers and firewalls in the call path; it is universally problematic in professional environments.