Expert assistance for Session Initiation Protocol (SIP) tasks including message construction, protocol analysis, security assessments, and VoIP pentesting. Use this skill whenever the user needs to understand SIP methods, create SIP messages, analyze SIP traffic, perform SIP security testing, work with SIP digest authentication, or troubleshoot VoIP systems. Trigger for any request involving SIP, VoIP, PBX, Asterisk, SIP headers, SIP response codes, or session initiation protocol.
Standardmäßig ist der Prompt ausgewählt, der zuerst die Quelle prüft. Sie können zu einem direkten Befehl wechseln oder eine lokale Kopie herunterladen.
Quelldateien prüfen
Lesen Sie SKILL.md und alle von SkillsMP angezeigten Begleitdateien, bevor Sie sich für eine Installation entscheiden.
Mit Codex oder Claude installieren Kopieren Sie diesen Prompt, fügen Sie ihn in Codex, Claude oder einen anderen Assistant ein und lassen Sie die Skill-Seite prüfen und installieren.
Ein direkter Befehl überspringt den Prüf-Prompt. Prüfen Sie die Quelle, bevor Sie ihn ausführen.
Expert assistance for Session Initiation Protocol (SIP) tasks including message construction, protocol analysis, security assessments, and VoIP pentesting. Use this skill whenever the user needs to understand SIP methods, create SIP messages, analyze SIP traffic, perform SIP security testing, work with SIP digest authentication, or troubleshoot VoIP systems. Trigger for any request involving SIP, VoIP, PBX, Asterisk, SIP headers, SIP response codes, or session initiation protocol.
SIP Protocol Assistant
A comprehensive skill for working with Session Initiation Protocol (SIP) - the signaling protocol used for establishing, modifying, and terminating multimedia sessions over IP networks.
What This Skill Does
This skill helps you:
Construct and parse SIP messages (INVITE, REGISTER, BYE, OPTIONS, etc.)
Understand SIP response codes and their meanings
Perform SIP security assessments and pentesting
Calculate SIP digest authentication responses
Analyze SIP traffic and troubleshoot VoIP issues
Harden SIP deployments against common vulnerabilities
Core SIP Methods
Request Methods
Method
Purpose
RFC
INVITE
Initiate or modify a session
3261
ACK
Confirm receipt of final INVITE response
3261
BYE
Terminate an established session
3261
CANCEL
Cancel a pending INVITE
3261
OPTIONS
Query server capabilities
3261
REGISTER
Register location with registrar
3261
SUBSCRIBE
Request event notifications
6665
NOTIFY
Send event notifications
6665
REFER
Request call transfer
3515
MESSAGE
Send instant messages
3428
UPDATE
Modify session without dialog state change
3311
PUBLISH
Publish event state to server
3903
Response Code Categories
Category
Range
Meaning
1xx
100-199
Provisional (request received, processing)
2xx
200-299
Success (request fulfilled)
3xx
300-399
Redirection (further action required)
4xx
400-499
Client Error (bad syntax or cannot fulfill)
5xx
500-599
Server Error (server failed valid request)
6xx
600-699
Global Failure (cannot fulfill anywhere)
Common Response Codes
100 Trying - Request received, processing
180 Ringing - Callee being alerted
183 Session Progress - Call progress information
200 OK - Request successful
301 Moved Permanently - Resource has new URI
302 Moved Temporarily - Resource temporarily at different URI
400 Bad Request - Malformed request
401 Unauthorized - Authentication required
403 Forbidden - Server refuses request
404 Not Found - Resource not found
408 Request Timeout - No complete request in time
486 Busy Here - Callee is busy
500 Internal Server Error - Server processing error
503 Service Unavailable - Server overloaded/maintenance
600 Busy Everywhere - All destinations busy
603 Decline - Callee refuses to participate
SIP Message Structure
Request Format
METHOD sip:uri SIP/2.0
Via: SIP/2.0/UDP/TCP/TLS host:port;branch=z9hG4bK...
Max-Forwards: 70
From: <sip:user@domain>;tag=...
To: <sip:user@domain>
Call-ID: unique-id@host
CSeq: sequence METHOD
Contact: <sip:user@host:port>
[Optional headers...]
Content-Type: application/sdp
Content-Length: bytes
[SDP body for INVITE/ACK]