DIDHub Telecom & VoIP Glossary
Plain-English definitions of every SIP, VoIP, telecom, and AI-voice term you are likely to encounter when working with phone numbers, SIP trunks, PBX systems, and STIR/SHAKEN. Each term has its own page with example messages, RFC links, and common pitfalls.
Trust & Security
STIR/SHAKEN
STIR/SHAKEN is a SIP caller-ID authentication framework that cryptographically signs every outbound call with an attestation level. It is mandated by the FCC for all originating US carriers and is ...
Attestation Levels (A, B, C)
Attestation levels are the three trust ratings that an originating carrier assigns to outbound calls under STIR/SHAKEN. They tell the terminating carrier how confident the originator is that the ca...
RFC 8224 (Authenticated Identity in SIP)
RFC 8224 is the IETF specification that defines the Identity SIP header for cryptographically signing the asserted caller-ID. It is the foundation of STIR/SHAKEN.
KYC (Know-Your-Customer) in Telecom
KYC is the process by which a telecom carrier verifies the identity, address, and authorization of a customer before issuing a phone number. KYC requirements vary by country — some require a ...
E911 (Enhanced 911) and the Laws Behind It
E911 is the Enhanced 911 emergency-services system in North America: dial 911, get connected to the nearest Public Safety Answering Point (PSAP), and have the dispatcher automatically see your call...
CNAM (Caller-ID Name)
CNAM is the human-readable name displayed alongside an inbound caller-ID. It is stored in distributed databases (LIDB) operated by US carriers and is queried at call-setup time, not during the SIP ...
Real-time / Web
WebRTC (Web Real-Time Communications)
WebRTC is a browser-native API and protocol stack for peer-to-peer audio, video, and data transfer. It runs in every modern browser without plugins and is the foundation for browser-based softphone...
SIP over WebSocket (WSS)
SIP over Secure WebSocket (WSS) is a transport binding (RFC 7118) that lets browsers and JavaScript softphones speak SIP signaling natively. Combined with WebRTC media, it is how zero-install brows...
DTLS-SRTP (Encrypted Media Keying)
DTLS-SRTP is the standard mechanism (RFC 5763) for negotiating SRTP encryption keys between two media endpoints over a DTLS handshake. It is the only keying method WebRTC supports, and it is the do...
ICE / STUN / TURN
ICE (Interactive Connectivity Establishment, RFC 8445) is a NAT-traversal protocol that finds a working media path between two endpoints by gathering candidate IPs/ports and trying them in priority...
NAT Traversal in SIP/VoIP
NAT (Network Address Translation) breaks SIP because SIP signaling carries IP addresses inside the message body (the SDP). NAT traversal techniques rewrite those addresses or use helper protocols s...
SRTP (Secure RTP)
SRTP (RFC 3711) is the encrypted version of RTP. It uses AES-CTR for confidentiality and HMAC-SHA1 for authentication, applied to the RTP payload. Without SRTP, anyone on the network path can recor...
SDES (a=crypto SRTP keying)
SDES (Session Description Protocol Security DEScriptions, RFC 4568) is the legacy method for negotiating SRTP encryption keys: the master key is base64-encoded directly into the SDP body via an
SIP Fundamentals
SIP INVITE
INVITE is the SIP method that initiates a call (or a media session). It carries the SDP that describes the offered codecs and media addresses, and it triggers the full request/response sequence tha...
SIP OPTIONS
OPTIONS is the SIP method used to query the capabilities of a remote endpoint — or, more commonly, as a keepalive 'ping' to check whether a SIP peer is still alive. It is the standard way car...
SIP REGISTER
REGISTER is the SIP method by which a user agent or PBX tells a SIP registrar 'I am at this IP address right now — route traffic for sip:user@domain to me.' It is the standard authentication ...
SIP Response Codes (1xx-6xx)
SIP response codes are 3-digit status codes patterned after HTTP. The first digit indicates the class: 1xx provisional, 2xx success, 3xx redirection, 4xx client error, 5xx server error, 6xx global ...
SDP (Session Description Protocol)
SDP (RFC 8866) is the body format inside a SIP INVITE that describes the offered media: codecs, IP addresses, ports, encryption, DTMF mode. It uses a simple line-oriented key=value syn...
RTP and RTCP
RTP (RFC 3550) carries the actual audio/video payload between endpoints. RTCP is the companion protocol that reports stats (packet loss, jitter, round-trip time) on the same media stream. Both run ...
Common SIP Headers (Explained)
Every SIP message is a list of headers (one per line) followed by an optional body (typically SDP). A handful of headers appear in nearly every request and reply — understanding them is the d...
SIP Authentication (Digest vs IP-ACL vs mTLS)
SIP authentication is how a server proves a SIP request really comes from an authorized client. Digest authentication (RFC 8760, formerly RFC 2617/7616) is the SIP standard, but real-world deployme...
SIP Transports (UDP / TCP / TLS / WSS)
SIP can run over four transports: UDP, TCP, TLS, and WebSocket Secure (WSS). They differ in reliability, encryption, NAT behavior, and latency. Picking the wrong transport is one of the most common...
IPv4 vs IPv6 in SIP / VoIP
SIP and RTP both run over IP, so IPv4 vs IPv6 is a real deployment choice — not a corner case. SIP carries IP addresses inside its message bodies (Via, Contact, SDP c= line), so IP-version mi...
Codecs & Media
G.711 (mu-law and A-law)
G.711 is the lowest-common-denominator narrowband audio codec on the PSTN. It is uncompressed PCM at 8 kHz sample rate, 8-bit logarithmic samples, yielding 64 kbps bitrate. Two regional flavors: mu...
G.729
G.729 is a narrowband audio codec at 8 kbps — one-eighth the bandwidth of G.711, useful for satellite, mobile, or constrained-network links. Patent royalties expired in 2017, so it is now roy...
Opus
Opus (RFC 6716) is a low-latency, highly adaptive audio codec covering bitrates from 6 kbps narrowband speech to 510 kbps fullband stereo music. It is the default codec for WebRTC and the strongly ...
T.38 (Fax over IP)
T.38 is the ITU-T standard for transporting real-time fax over IP networks. It does not transmit fax audio — it relays the underlying T.30 fax protocol over UDPTL or TCP, which is far more re...
DTMF (RFC 4733 / RFC 2833)
DTMF (Dual-Tone Multi-Frequency) is the touch-tone signaling used in IVRs, conference bridges, and call menus. RFC 4733 (formerly RFC 2833) defines how to carry DTMF events out-of-band in ...
Numbering
E.164
E.164 is the ITU-T international public-telecom numbering plan. It defines the format used worldwide: a leading + followed by a country code (1-3 digits) and a national number, max 15 ...
DID (Direct Inward Dialing)
A DID is an externally-dialable phone number that routes inbound calls directly to a specific endpoint (extension, IVR, queue, AI agent) on a PBX or SIP trunk — without going through a switch...
Toll-Free Number
A toll-free number is a phone number where the called party (not the caller) pays for the call. In the US, toll-free uses NPAs 800, 833, 844, 855, 866, 877, 888 (and the legacy 880-887). Most other...
OBR (Origin-Based Rating)
OBR is a wholesale telecom pricing model where outbound call rates depend on where the call originated from (the calling number's country) as well as where it is going. It applies...
Architecture
PBX (Private Branch Exchange)
A PBX is a private telephone switch that routes calls between extensions inside an organization and connects to outside lines (PSTN trunks) for external calls. In the legacy AT&T switching hier...
SBC (Session Border Controller)
An SBC sits at the edge between a private network (PBX, internal SIP) and an external SIP/PSTN network (carrier, internet). It handles security, NAT traversal, codec transcoding, topology hiding, a...
BYOC (Bring Your Own Carrier)
BYOC is a deployment model where you use a third-party SaaS platform (Vapi, Retell, Microsoft Teams, Zoom Phone, Twilio Flex) for the call-control / AI-agent / contact-center logic, but route the a...
SIP Trunk
A SIP trunk is the carrier-side endpoint that connects your PBX (or AI voice agent, or softphone) to the PSTN over SIP. It is the modern replacement for the analog or T1/E1 'trunk' that PBXs used t...
Microsoft Teams Direct Routing
Direct Routing is Microsoft's BYOC mechanism for Teams Phone — it lets you bring your own SIP carrier instead of buying Microsoft's bundled Calling Plans. You connect Teams to your carrier th...
Class 4 vs Class 5 Switches (Trunking vs PBX)
Class 4 and Class 5 are inherited from the original AT&T 1950s switching hierarchy, but the distinction still defines how modern softswitches, SIP trunking carriers, and PBXs are built. Class 4...
Auto-Provisioning (zero-touch desk phone setup)
Auto-provisioning is how you deploy 50, 500, or 50,000 desk phones without manually configuring each one. The phone boots, fetches its config from a central server using its MAC address, applies it...
Desk Phones vs Softphones
A desk phone is a dedicated hardware SIP endpoint — Polycom, Yealink, Snom, Cisco. A softphone is software running on a computer, mobile, or browser — Microsoft Teams app, Zoiper, Linph...
Mobile SIP Apps & the Background-Running Challenge
Running a SIP softphone on a phone (iOS or Android) sounds simple but is fundamentally hard. Mobile OSes aggressively suspend background apps to save battery, which kills SIP REGISTER refreshes and...
VoIP vs PBX (and the difference)
VoIP and PBX are often used interchangeably but mean different things. VoIP is the transport layer (calls over IP). PBX is the application layer (call control, extensions, IVR, re...
Software & Stacks
Asterisk (open-source PBX framework)
Asterisk is the original open-source telephony framework, started by Mark Spencer in 1999. It is a Class 5 PBX engine: it terminates SIP/IAX/PJSIP endpoints, runs dialplan logic, plays audio, recor...
FreeSWITCH
FreeSWITCH is an open-source telephony engine started by Anthony Minessale in 2006 (originally as a multi-threaded successor to Asterisk). It is more flexible than Asterisk — it can be used a...
Kamailio (SIP server / softswitch)
Kamailio (formerly OpenSER) is a high-performance SIP server. It is not a PBX — it has no media, no IVR, no voicemail. It is a SIP signaling proxy/router/registrar that handles tens ...
OpenSIPS (SIP server / fork of OpenSER)
OpenSIPS is a high-performance SIP server forked from OpenSER in 2008 (the same project Kamailio also forked from). Like Kamailio, it is a Class 4 SIP signaling engine — routing, registration...
rtpengine (media proxy / relay)
rtpengine is an open-source kernel-accelerated RTP proxy maintained by Sipwise. Kamailio and OpenSIPS handle SIP signaling but not media; rtpengine is the standard companion that handles the actual...
coturn (open-source STUN/TURN server)
coturn is the standard open-source STUN/TURN server. It is the relay-of-last-resort that lets WebRTC and SIP-over-NAT clients reach each other when direct connectivity fails. If you run any browser...
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