SIP
SIP (Session Initiation Protocol) is the standard signalling protocol for setting up, modifying, and terminating voice/video sessions over IP. In AV, it is how room DSPs, room codecs, UC appliances, intercom gateways, paging systems, door stations, softphones, and similar endpoints register with or call through a PBX, hosted voice platform, cloud calling service, conference bridge, or SBC. SIP handles signalling only: registration, call setup, routing, capability negotiation, mid-call changes, keepalives, and teardown. The actual media travels separately via RTP / RTCP, optionally secured as SRTP, on implementation-specific ports. NAT traversal, DNS/profile mismatch, certificate identity, and media-range ownership remain the most common real deployment failures.
Scope: SIP — select the deployed product/version; optional services require separate selection.
protocol · Catalog reviewed 2026-10-21
SIP, IETF, ITU, IEEE
Related requirements
Review these entries for the services used in your design.
Infrastructure Requirements
- A defined SIP role model (user agent, proxy, registrar, redirect server, PBX, hosted voice service, B2BUA, SBC, direct peer, or mobile/client push-notification edge)
- chosen signalling transport (UDP, TCP, TLS, SCTP, or WebSocket where explicitly supported)
- the real media path using RTP / RTCP or SRTP
- SDP / offer-answer compatibility
- codec and DTMF method alignment
- DNS NAPTR / SRV / A / AAAA behavior if the design uses domains
- NAT traversal by SBC, ICE, STUN, TURN, or a controlled private path
- and certificate / FQDN hygiene when TLS is used
Network & Physical Requirements
| Port / allocation | Transport | Connection / signal flow | Feature and requirement | Evidence |
|---|---|---|---|---|
| 5060 (default) | TCP/UDP | User agent / proxy → SIP peer / registrar | Profile requirement · Core SIP request/response signalling SIP RFC 3261 | Source checked 2026-09-27 RFC 3261, Section 18.1.1: Sending Requests (default transport ports) |
| 5061 (default) | TCP | User agent / proxy → TLS SIP peer | Optional · TLS signalling SIP over TLS SIPS / TLS transport | Source checked 2026-09-27 RFC 3261, Section 18.1.1: Sending Requests (default transport ports) |
| Negotiated | UDP | Media sender → Media receiver | Profile requirement · Core Negotiated RTP and RTCP media SDP-negotiated RTP/RTCP Use the PBX and endpoint-configured media range; signalling ports do not cover media. | Source checked 2026-09-27 RFC 3550, Section 11: RTP over Network and Transport Protocols |
Gotchas & IT Notes
- ⚠Match registrar/proxy, authentication, codecs, DTMF and NAT/traversal policy. TCP, UDP, TLS and WebSocket support depend on both ends.
- ⚠SRTP is separate from signalling TLS. Restrict actual media ranges and destinations from SDP and endpoint configuration.