SIP does not replace workforce management, quality monitoring, CRM, emergency policy, or a messaging registry. It is the session seam those systems hang from. Compliance notes on this page are educational and are not legal advice.
Educational only. Not legal advice, not a compliance opinion, and not a filing. Statutes and the pages linked from each card outrank this paraphrase.
{"content":"SIP session","children":[{"content":"Signaling","children":[{"content":"INVITE and responses"},{"content":"WebSocket subprotocol sip"},{"content":"Transactions and dialogs"}]},{"content":"Media","children":[{"content":"SDP offer and answer"},{"content":"ICE, STUN, TURN"},{"content":"DTLS-SRTP and Opus"}]},{"content":"Open core","children":[{"content":"Kamailio SIP server"},{"content":"Asterisk and FreeSWITCH"},{"content":"rtpengine"}]},{"content":"Compliance study","children":[{"content":"STIR and SHAKEN"},{"content":"Kari's Law and RAY BAUM'S Act"},{"content":"10DLC messaging"}]},{"content":"Neighbors","children":[{"content":"WFM and CRM"},{"content":"SIPREC recording"},{"content":"Programmable voice APIs"}]}]}WFM forecasts volume and builds schedules. It consumes contact-center events. It does not parse Via headers. When a forecast is wrong, the SIP symptom is usually queue delay and 480 or carrier congestion, not a broken dialog ID.
QM scores interactions. The standardized way an SBC forks a session to a recorder is SIPREC.
Location has to ride with the call. RFC 6881 is the BCP, RFC 6442 carries location in SIP, and the FCC MLTS page states Kari’s Law and RAY BAUM’s Act obligations for multi-line systems.
Amazon Connect can attach conversational analytics to an external voice system with a SIPREC connector. The analytics plane is not a second SIP proxy for the customer call.
Screen pop uses data the contact center already has: ANI, DNIS, case id. Those values may arrive in SIP headers or in a CTI event after the call is up. The CRM is not a SIP element.
A voice bot still needs a session. On Amazon Connect the platform owns the managed PSTN voice channel. Bridging an outside carrier into Connect with a Chime Voice Connector is not a supported pattern unless AWS reviews it. The FreeSWITCH manual page also describes SignalWire, founded by the FreeSWITCH team, as a hosted platform with AI agents behind REST APIs and SWML.
Twilio’s docs home is API reference, tutorials, and integration for Twilio products. Elastic SIP Trunking is the SIP-shaped product already cited in this lab. This page does not define the word CPaaS from an RFC. Read the vendor page before you copy a diagram that uses that word.
CUCM is the enterprise registrar and the SIP trunk endpoint toward CUBE or another SBC. A CUCM SIP Profile is an admin object. It is not a CUBE voice class sip-profiles list.
UCCX call control is through CUCM. Carrier SIP does not terminate on UCCX. Confirm the solution design guide for the version you run.
CVP configuration guides describe SIP between the call server, gateways, and CUCM. Use the guide that matches the version. This lab does not invent a script.
Unity Connection’s documented SIP integration sits on a CUCM SIP trunk. Open the integration guide for your Connection version before you copy a trunk setting.
Premises PSTN uses a Local Gateway. Cloud Connected PSTN is a partner PSTN service. They are different location settings.
This is a separate product from Webex Calling. Do not assume the Local Gateway pattern applies. Use Cisco’s current design guide for that product.
Direct Routing uses a certified SBC, TLS to the pstnhub FQDNs, and SRTP media. Users are homed in Microsoft 365.
Genesys documents BYOC Cloud and BYOC Premises. Five9 documents BYO SIP to its SBCs. Twilio documents Elastic SIP Trunking. NICE documents connectivity ports and leaves carrier design account-specific.
Kamailio’s home page describes an open source SIP server that can front a PBX such as Asterisk or FreeSWITCH. rtpengine’s README describes the RTP proxy meant to sit with Kamailio. A CUCM SIP trunk toward that SIP server is still SIP. The CUCM SIP Profile and Asterisk’s pjsip.conf are different objects.
chan_pjsip depends on res_pjsip. Accounts go in pjsip.conf. The dialplan is extensions.conf. Open the examples for the Asterisk version you run.
The users manual covers SIP profiles, the dialplan, codecs, media, WebRTC, and the Event Socket. It says its parameter tables were checked against the source. This lab does not copy those defaults.
Use Kamailio for the SIP server roles its home page lists. Use rtpengine when SDP must be rewritten or when the README’s ICE and SRTP bridges are required. GNU/Linux is the platform that README names.
RFC 7118 requires the WebSocket subprotocol sip. Via uses WS or WSS. The URI transport parameter the RFC adds is transport=ws. A sips path that includes WebSocket must be a secure WebSocket.
Stop at the WebSocket handshake. A missing 101 means the SIP request was never framed. Codecs are not negotiated yet.
Read c= and m=, then ICE candidates, then DTLS-SRTP. A private address in c= is a reachability problem. Identity headers do not move RTP.
RFC 8224 defines the Identity header. RFC 8225 defines PASSporT. The STI-GA page describes SHAKEN. A placeholder token is not a verification. Not legal advice.
Public Law 115-127 is the direct-dial statute. Section 506 of RAY BAUM’S Act is the dispatchable-location proceeding. RFC 6442 is SIP location conveyance. Do not place a live 911 call for this lab. Not legal advice.
10DLC on The Campaign Registry site is a messaging channel. Do not debug it as a missing rtpmap line.
Control protocols named on the Asterisk license page. The page states Digium’s belief about applications that use them. This lab does not paste an ARI request and does not give license advice.
The FreeSWITCH manual’s programming interface for the server you run. SWML on that same page is the hosted platform, not the local dialplan file.
RFC 7866 is the recording session. It is a second SIP dialog, not a CRM webhook.
RFC 7118 is the browser signaling hook when the application chooses SIP. JSEP also allows a custom protocol.
A mind-map box is not an API. If this lab did not open the vendor page, the sources block says confirm and prints no URL.