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SIP UniverseFree University

Glossary

Terms

SIP

Session Initiation Protocol. Application signaling that creates, modifies, and ends sessions.

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SDP

Session Description Protocol. The text format that describes media streams for offer/answer.

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Offer/answer

The rule set that pairs one SDP offer with one SDP answer. The answer mirrors the offer’s m-line count and order.

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RTP

Real-time Transport Protocol. Carries media samples. It is not SIP.

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RTCP

RTP Control Protocol. Reports loss, jitter, and timing alongside RTP.

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SRTP

Secure RTP. Encrypts and authenticates media packets.

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SDES

Security Descriptions for SDP. Puts SRTP keying material in a=crypto.

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DTLS-SRTP

Keying SRTP with a DTLS handshake instead of embedding keys in SDP.

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SBC

Session border controller. A deployment role, usually a B2BUA, sitting on a trust boundary. Requirements are collected in RFC 5853.

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B2BUA

Back-to-back user agent. A UAS on one side glued to a UAC on the other. Taxonomy: RFC 7092.

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User agent

The SIP endpoint that originates or answers requests.

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UAC

User agent client. The side that sends a request.

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UAS

User agent server. The side that answers a request.

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Proxy

A SIP element that routes requests and responses without, by itself, being the session endpoint.

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Registrar

A server that accepts REGISTER and stores Contact bindings.

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Redirect server

A server that answers with 3xx and does not stay in the path.

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Transaction

One request and its responses, identified on a hop by the Via branch.

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Dialog

A peer relationship identified by Call-ID, local tag, and remote tag.

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Session

The media relationship negotiated with offer/answer. Distinct from the SIP dialog.

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Call-ID

Header that stays constant for a dialog. Compact form: i.

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From tag

The tag parameter the UAC puts on From. It becomes part of the dialog ID.

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To tag

The tag the UAS adds to To. Absent on the initial request.

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CSeq

Command sequence: a number and a method token. The method must match the request.

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Via

The hop the response must return through. Compact form: v. The branch parameter identifies the transaction.

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Magic cookie

The branch prefix z9hG4bK required by RFC 3261 so proxies can tell old branches from new ones.

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Max-Forwards

Hop limit on a request. The recommended initial value is 70.

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Contact

Where this UA wants to be reached for this dialog. Compact form: m.

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Record-Route

A proxy’s request to stay on the path. The route set is built from these values.

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Route

The path a UA sends an in-dialog request along, learned from Record-Route.

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Request-URI

The URI on the request line. It is the current target, not always the original To URI.

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SIP URI

sip:user@host form defined by RFC 3261.

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SIPS URI

sip over TLS scheme. It is not itself an SRTP promise.

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tel URI

A telephone-number URI.

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E.164

The ITU numbering plan. SIP carries those digits; it does not define the plan.

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Early offer

An INVITE that already contains the SDP offer.

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Delayed offer

An INVITE with no SDP. The offer arrives in a reliable response.

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PRACK

Provisional acknowledgement. Makes selected 1xx responses reliable when 100rel is in use.

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100rel

The option tag that turns on reliable provisional responses.

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UPDATE

A method that can change session parameters, including before the initial INVITE finishes.

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re-INVITE

A mid-dialog INVITE used to modify the session or refresh the target.

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ACK

Acknowledges a final response to INVITE. The 2xx ACK is end to end; the non-2xx ACK is hop by hop.

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BYE

Ends an established dialog.

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CANCEL

Gives up on an INVITE that has not received a final response.

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OPTIONS

Asks a peer what it supports, and is widely reused as a keepalive.

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REGISTER

Creates or refreshes a binding between an address-of-record and a Contact.

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REFER

Asks another UA to issue a request, typically for transfer.

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Replaces

Header that identifies a dialog to be replaced, used with attended transfer.

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INFO

Mid-dialog application information. Not the IETF way to send DTMF digits.

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KPML

Keypad markup for reporting digits in the signaling plane.

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telephone-event

RTP payload for DTMF and related tones.

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Session-Expires

Header that proposes how long a session may stay up without a refresh.

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422

Session Interval Too Small. Defined with session timers, not in the original response list as a timer code.

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488

Not Acceptable Here. This UA cannot accept the offered media.

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487

Request Terminated. The usual final response after a successful CANCEL.

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407

Proxy Authentication Required. Answer with Proxy-Authorization.

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401

Unauthorized. Answer with Authorization. This is the UAS challenge, not the proxy one.

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GRUU

Globally Routable User Agent URI. A Contact that reaches one instance.

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Outbound

RFC 5626 flow management: reg-id, sip.instance, and keepalives so a NAT binding stays up.

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Path

Header a registrar uses to remember the edge proxies that must stay on a registration.

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Service-Route

Header a registrar returns so the UA learns the path for outbound initial requests.

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Loose routing

A Route/Record-Route URI with the lr parameter. The request URI stays the target.

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rport

Via parameter that tells the peer to answer the source port it actually saw.

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STUN

Session Traversal Utilities for NAT. Binding checks and mapped addresses.

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TURN

Traversal Using Relays around NAT. Allocates a relayed media address.

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ICE

Interactive Connectivity Establishment. Tries candidate pairs until one works.

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SIP ALG

A middlebox that rewrites SIP or SDP addresses. It is a common cause of one-way audio and it cannot rewrite TLS.

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PAI

P-Asserted-Identity. An identity asserted inside a trusted network, not a cryptographic proof.

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PPI

P-Preferred-Identity. What an untrusted UA would like the trusted network to assert.

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Privacy

The Privacy header and related behavior for withholding identity.

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STIR

Secure Telephone Identity Revisited. Cryptographic Identity header for SIP.

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PASSporT

The token format STIR carries.

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SHAKEN

The telephone-industry profile for using STIR on the PSTN. Attestation details live in ATIS specifications; the FCC describes the program.

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  • FCC call authentication — confirm

Digest authentication

Challenge-response using 401 or 407, specified for SIP by reference to HTTP digest.

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SIPREC

SIP recording protocol. A session is forked to a recording server.

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SRS

Session recording server. The SIPREC receiver.

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E911

Emergency calling from multi-line systems. Location and routing rules are in RFC 6881 and FCC MLTS rules.

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Service URN

A URN such as urn:service:sos that names an emergency service.

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EF

Expedited Forwarding per-hop behavior, the diffserv class RFC 4594 maps voice toward.

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ptime

SDP attribute for packetization time, in milliseconds.

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rtpmap

SDP attribute that names the codec for a payload type.

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fmtp

SDP attribute that carries codec parameters.

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G.722 clock

RFC 3551 assigns G.722 the rtpmap rate 8000 even though the audio is 16 kHz.

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SIP profile

On CUBE, a voice class sip-profiles ruleset that rewrites SIP or SDP. Not the same object as a CUCM SIP Profile.

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Local Gateway

The customer-premises SBC role in Webex Calling premises-based PSTN.

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Cloud Connected PSTN

A Webex Calling PSTN option delivered by a Cisco PSTN partner, not by a Local Gateway.

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Direct Routing

Microsoft Teams Phone connectivity through a certified customer SBC.

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Media bypass

A Teams Direct Routing option described in Microsoft’s planning articles. Read the current article before you enable it; this glossary does not restate its conditions.

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BYOC

Bring your own carrier. In Genesys Cloud this is a documented product name for BYOC Cloud and BYOC Premises.

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Elastic SIP trunk

Twilio’s name for a SIP trunk with separate termination and origination settings.

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Termination

In Twilio Elastic SIP Trunking, calls from your PBX toward the PSTN.

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Origination

In Twilio Elastic SIP Trunking, calls from the PSTN toward your PBX.

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Early media

Media that flows before the call is answered, discussed in RFC 3960.

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One-way audio

Signaling succeeds and RTP flows in only one direction. Start with c=, m=, and the return path.

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Toll fraud

Unauthorized calling that creates carrier charges. Prevention on IOS voice includes the trusted IP list.

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Content-Length

Byte length of the body. Compact form: l. A mismatch means the framing is wrong.

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Compact form

Single-letter header names defined for a subset of headers.

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Max-Forwards exhaustion

A request that arrives with Max-Forwards 0 is answered with 483 Too Many Hops.

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Forking

A proxy sending one request to several targets, which can create early dialogs.

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Early dialog

A dialog created by a provisional response that already carries a To tag.

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Target refresh

Changing the remote target with a mid-dialog request. re-INVITE handling is covered by RFC 6141.

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History-Info

A standards-track record of request history.

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Diversion

A widely implemented, non-standards-track header for forwarding history.

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Reason

A header that carries why a request or response happened.

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WebSocket transport

SIP over WebSocket, used by browser user agents.

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Domain certificate

Guidance for which names belong on a SIP TLS certificate.

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Connection reuse

RFC 5923 aliasing. Microsoft’s Direct Routing proxy documentation says it does not support this aliasing.

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Documentation address

203.0.113.0/24, 198.51.100.0/24, and 192.0.2.0/24 are reserved for examples.

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JSEP

JavaScript Session Establishment Protocol. The application chooses signaling. RFC 8829 names SIP as one option.

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SIP WebSocket subprotocol

The WebSocket subprotocol identifier sip, used to carry SIP after the opening handshake.

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Opus rtpmap

RFC 7587 requires the rtpmap clock 48000 and two channels: opus/48000/2.

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Kamailio

Open source SIP server under GPLv2+. The project page says it can front a PBX such as Asterisk or FreeSWITCH.

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Asterisk

Open source PBX. Current SIP configuration for chan_pjsip is res_pjsip in pjsip.conf, with the dialplan in extensions.conf.

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FreeSWITCH

Telephony server whose manual covers SIP profiles, the dialplan, media, WebRTC, and the Event Socket.

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rtpengine

Sipwise media proxy for RTP and other UDP media, meant to be used with Kamailio. GNU/Linux.

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RAY BAUM'S Act

A division of Public Law 115-141. Section 506 requires a proceeding on dispatchable location, including from MLTS. Not legal advice.

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Dispatchable location

In section 506, the street address of the calling party plus room, floor, or similar information needed to identify the caller.

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10DLC

On The Campaign Registry site, an application-to-person messaging channel. Not a SIP method. Not legal advice.

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ARI

Asterisk REST Interface. Named with AMI and AGI on the Asterisk license page as a control protocol.

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