VoIP & Messaging

WAV is the universal file format for call recordings, IVR prompts, voicemail exports, and PSTN debug captures. Every telephony platform (Twilio, Asterisk, FreeSWITCH) reads and writes WAV-wrapped G.711.

Codec mismatch debugging starts here — SDP says `PCMU/8000` which is PT 0 per RFC 3551. Static vs dynamic payload types explain most SIP/WebRTC negotiation confusion.

IVR navigation, call routing, and fax detection depend on RFC 4733 DTMF — not literal G.711 tone audio. Misconfigured `telephone-event` in SDP breaks keypad input on SIP calls.

Every phone number your app touches — Twilio, AWS SNS, WhatsApp, SIP — uses E.164. Getting number formatting and parsing right (libphonenumber patterns) requires understanding CC/NDC/SN structure.

SIP is the PSTN of the internet. Every VoIP platform (Twilio, Vonage, AWS Chime, Asterisk, FreeSWITCH) speaks SIP. Building any communications product requires understanding INVITE flows, registration, and codec negotiation via SDP.

SDP is the handshake language between any two endpoints setting up a call. WebRTC, SIP, and RTSP all use it. You can't debug codec mismatches, ICE failures, or video call setup without reading SDP.

RTP carries all real-time media: VoIP, WebRTC, live video, and video conferencing. Understanding SSRC demultiplexing, jitter buffers, and RTCP feedback loops is essential for any media application.

Any media you send over the internet should be encrypted. SRTP is the protocol. In WebRTC it's automatic via DTLS-SRTP; in SIP deployments you must configure it explicitly. Know it when auditing VoIP security.

Opus is the codec for real-time audio on the internet. WebRTC mandates it; Discord, Zoom, and most VoIP apps use it. Understanding bitrate, FEC, and frame sizing directly impacts audio quality and bandwidth.

SMS is how billions of people receive OTP codes, alerts, and notifications. Understanding PDU format, encoding (GSM-7 vs UCS-2), and concatenation is essential for any platform sending SMS at scale via Twilio, AWS SNS, or direct SMPP.

AMR-NB is the fallback voice codec for pre-VoLTE mobile networks and many carrier voicemail systems. The `#!AMR` file header and RFC 4867 RTP framing are distinct from AMR-WB.

3GP is the standard container for mobile-recorded video/voice messages, MMS media, and legacy carrier content. Shares ISO BMFF box structure with MP4 but restricted codec set.

HLS historically delivered MPEG-TS segments — still common in broadcast IPTV and legacy VoD. TS packet structure explains why HLS segment boundaries differ from fMP4 CMAF.

WebRTC and modern SIP use RFC 7587 Opus RTP framing — not Ogg pages. FEC and DTX SDP attributes directly control call quality under packet loss.

VoLTE and IMS carry AMR over RTP per RFC 4867 — not as `.amr` files. CMR bits and octet-align vs bandwidth-efficient modes cause real interoperability bugs between carriers.

SIP over WebSocket is how browser-based softphones and WebRTC-SIP gateways work. If you're building a web-based communications app that bridges to PSTN or SIP trunks, this is the transport spec.

G.711 is the common denominator of telephony. Every PSTN gateway and legacy SIP trunk speaks G.711. When codec negotiation fails and you're left with 64 kbps PCM, this is why.

RCS is replacing SMS for brand messaging, OTP, and consumer engagement. Every major US/EU carrier now supports RCS Universal Profile. If you build SMS communications today, you need to understand RCS UP for 2025+ deployments.

XMPP powers enterprise messaging (Cisco Jabber), IoT device management, and many open federated chat systems. WhatsApp internally uses a modified XMPP protocol for message delivery.

The roster and subscription model in RFC 6121 defines how federated chat contacts work. Essential for implementing XMPP clients, servers, or S2S (server-to-server) federation between XMPP deployments.

Matrix is the modern alternative to XMPP for federated messaging: bridges to Discord/Slack/WhatsApp, end-to-end encryption by default, and full decentralization. Used by the German government, French public sector, and Mozilla.

Legacy GSM voice and older Asterisk/voicemail systems still produce `.gsm` files. Distinct from the GSM cellular network spec — this is the audio codec bitstream format.

The AMR file header is a minimal envelope pattern — one magic line then framed payloads. Required when parsing `.amr`/`.awb` attachments from carrier APIs or MMS gateways.

G.729 remains common on legacy SIP/PBX trunks where bandwidth is constrained. License/patent history affected adoption — know it when debugging codec negotiation on older enterprise systems.

G.726 appears in older VoIP gateways and recorded WAV files. Often confused with G.711 — verify fmt chunk bit rate and RTP payload type mapping.

EVS is the next-generation carrier voice codec after AMR-WB. Required for 5G voice debugging, operator IMS integrations, and understanding VoNR codec policies.

iLBC remains in legacy WebRTC/SIP endpoints and some embedded VoIP devices. Frame independence (no inter-frame dependency) makes it resilient to packet loss — useful design comparison for Opus FEC.

Speex is Opus's predecessor from Xiph.Org. Understanding Speex/Ogg framing helps maintain legacy VoIP and streaming integrations.

Opus-in-Ogg is an alternative to WebM/MP4 for VoIP recordings and Icecast streams. Page/chunk structure is a classic telephony streaming envelope pattern.

Vorbis shares Ogg framing with Opus — understanding Vorbis header packets clarifies how Ogg-encapsulated telephony codecs are structured.

5G VoNR voice debugging requires RFC 8130 — EVS RTP framing differs substantially from RFC 4867 AMR. TOC parsing is essential for packet capture analysis.

G.711 with VAD, G.729B, and Opus DTX all rely on comfort noise signaling. Missing CN payloads cause 'dead air' perception and hold-music glitches in contact centers.

Telephony and messaging apps exchange contacts as `.vcf` files — parsed in MMS (MMS vCard attachments), QR contact cards, and SIP presence. TEL property formatting ties directly to E.164.

ENUM is how SIP networks look up where to route a call to a phone number. Essential for VoIP providers, number portability implementations, and understanding how PSTN-to-SIP calls are routed.

H.323 remains in enterprise video conferencing (Cisco, Polycom legacy endpoints) and some carriers. Understanding it is necessary when integrating with legacy infrastructure or gateways.

G.722 is the standard for HD voice in SIP and enterprise telephony. When you enable 'HD Voice' on a SIP trunk, G.722 is typically what's negotiated. Essential for enterprise UC deployments.

AMR-WB is mandatory for VoLTE — every LTE voice call uses it. Understanding rate adaptation and the AMR payload format (RFC 4867) is essential for mobile VoIP and operator codec policy.

MMS is still widely used for group messaging (iMessage fallback on non-Apple devices), marketing images, and vCards. Understanding MM7 (application-to-MMSC API) is needed when integrating with carrier MMS gateways.