Foundations and how to read a call
What SIP is for, which network facts it assumes, and how to read a ladder without inventing rules.
7 rooms
SIP study · 20 tracks · offline
Learn SIP in color.
From the first INVITE to borders, media, and interop. Every room has a five-minute path, a drawing, a worked example, a quiz, and sources.
132 rooms
792 prompts
528 cards
142 diagrams
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SIP core: messages, transactions, dialogs
Roles, methods, headers, transaction timers, dialog identity, and the transports those rules sit on.
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Call flows that operators actually trace
The ACK split, route sets, PRACK, UPDATE, session timers, teardown, and transfer.
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Registration, DNS, and routing
Bindings, DNS location, loose routing, outbound flows, Path, Service-Route, GRUU, and forking.
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SDP and offer/answer
How to read an SDP body, and how RFC 3264 matches offers to answers without reordering streams.
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RTP, codecs, and DTMF
RTP headers, payload types, codec documents, RTCP, telephone-event, packet time, and one-way audio.
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TLS, SRTP, identity, and abuse
Defensive digest, TLS certificates, SRTP keying, privacy headers, STIR, and abuse controls.
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NAT, ICE, and the ALG problem
Why embedded addresses break, and what rport, STUN, TURN, ICE, SIP ALGs, and media latching each actually do.
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Session border controllers
Deployment requirements, the B2BUA role, and the functions whose limits stay on the vendor datasheet.
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Cisco CUBE, CUCM, and Unity Connection
CUBE as an IOS XE B2BUA, the dial-peer and SIP profile tools, and the CUCM and Unity documents that use different objects with similar names.
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Oracle, Ribbon, and AudioCodes
Object names and manipulation syntax you can verify in public docs, and the certificate name the peer actually checks.
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Cloud voice and CCaaS SIP
Documented SIP boundaries for Teams, Webex Calling, Amazon Connect, Five9, Twilio, Genesys, and NICE CXone.
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Troubleshooting ladders, captures, and debug
Match Call-ID, CSeq, Via branch, and To tag, then read captures and IOS debug as themselves.
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Toll fraud, E911, recording, and QoS
Defensive operations: source restrictions, emergency location, disclosed SIPREC, and the EF PHB.
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Ecosystem, certifications, and the UC bridge
Name only the certifications this edition confirmed, and keep CUCM, UCCX, CVP, and Webex Calling in their real roles.
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Interviews, incidents, and design drills
Cited answers in the lesson body, one-variable incidents, and interconnect checklists.
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WebRTC and SIP over WebSocket
How a browser reaches SIP: RFC 7118 on the signaling socket, ICE for candidates, and DTLS-SRTP for the media.
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Open-source telephony core
Asterisk, FreeSWITCH, Kamailio, and rtpengine do different jobs. The pages below say which job each project claims.
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Call authentication and compliance
STIR/SHAKEN, Kari's Law, RAY BAUM'S Act, and 10DLC, as the opened documents describe them. Educational only. Not legal advice.
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Ecosystem, maps, and integration hooks
Glossary, sister systems, configuration, bridges from enterprise UC to the open projects, and the hooks those projects actually name.
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T01 Ports, transports, and where a request can go
T01 E.164, sip URIs, and tel URIs
T01 The hitchhiker guide is a map
T01 Licenses, codecs, and what you may copy
T02 Roles a trace is actually showing
T02 Transactions and the branch cookie
T02 T1, the transaction timeout, and Timer C
T02 UDP, TCP, TLS, and WebSocket
T03 Record-Route and the route set
T03 PRACK and reliable provisionals
T03 UPDATE before the INVITE completes
T03 CANCEL, BYE, and re-INVITE
T04 Locating a server with RFC 3263
T04 Loose routing and Max-Forwards
T04 Outbound, reg-id, and keepalives
T04 GRUU and a single instance
T05 rtpmap names and fmtp parameters
T05 Direction attributes and hold
T05 Offer/answer pitfalls in RFC 6337
T06 Static and dynamic payload types
T06 RTCP sender and receiver reports
T06 telephone-event and other digit paths
T06 ptime, packet rate, and EF
T07 TLS and domain certificates
T07 SRTP and two keying methods
T07 Privacy and asserted identity
T07 STIR, PASSporT, and SHAKEN
T07 Scanning, hijack, and toll fraud
T08 Embedded addresses and NAT
T08 rport and symmetric responses
T08 STUN server-reflexive addresses
T09 RFC 5853 is requirements, not a protocol
T09 An SBC is commonly a B2BUA
T09 Topology hiding rewrites the addresses you leak
T09 Anchor, bypass, transcode, and DTMF
T09 Call admission stays on the datasheet
T09 Active and standby, without a made-up timer
T10 Dial peers select the SIP next hop
T10 Early offer and delayed offer
T10 CUCM trunks, SIP profiles, and Lua
T10 Unity Connection SIP integration
T11 Ribbon Edge, Core, and public validation lists
T11 AudioCodes manipulation in software 7.6
T11 Three manipulation grammars
T11 Identity at a trust boundary
T11 Certificate names and Contact
T12 Teams Direct Routing signaling
T12 OPTIONS health and SRTP media
T12 Webex Calling Local Gateway
T12 Three PSTN choices, one per location
T12 Amazon Connect voice boundaries
T12 Five9 BYO SIP and Twilio trunks
T12 Genesys BYOC and NICE CXone SIP
T13 Match a ladder before you change anything
T13 Status codes you will actually see
T13 Display filters for SIP and RTP
T13 488 codecs and 422 session timers
T13 TLS names, trust anchors, and version
T13 Read an RFC header before you cite it
T14 The toll-fraud class, stated defensively
T14 Trusted addresses and source restrictions
T14 Emergency calling, location, and the FCC page
T14 Disclosed recording with SIPREC
T14 Voice, EF, and what the PHB does not design
T14 Alarms that match the failure
T15 A glossary tied to a document
T15 Neighbors that are not SIP elements
T15 Workforce tools and agents above the media
T15 Certifications this edition will name
T15 When the page and the memory disagree
T15 CUCM, UCCX, and CVP without a fake script
T15 Webex Calling local gateway, and a separate contact center
T16 Signaling drill with the cited answer
T16 Media drill with the cited answer
T16 SBC drill with the cited answer
T16 Symptom, ladder, one packet, one variable
T16 An interconnect checklist, not a config dump
T16 CUCM, a certified SBC, and Teams Direct Routing
T17 Signaling, JSEP, and the media plane
T17 RFC 7118: the sip subprotocol
T17 DTLS-SRTP and the required audio codecs
T17 Browser softphone call flow
T18 Asterisk, PJSIP, and the dialplan
T18 FreeSWITCH profiles, dialplan, and WebRTC
T18 Kamailio as the SIP server
T18 rtpengine and the integration map
T19 How to read a compliance page
T19 Kari's Law, RAY BAUM'S Act, and SIP location
T19 10DLC is a messaging channel
T19 Troubleshooting without a legal conclusion
T20 Sister technologies around the session
T20 Configuration and troubleshooting seams