VoIP & Messaging

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.

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.

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.