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- Create Date March 13, 2017
- Last Updated September 16, 2026
A Session Initiation Protocol (SIP) Profile for VoIP analysis in Wireshark
This profile is for VoIP SIP protocol analysis in Wireshark. You need a number of profiles to be successful with complete VoIP packet analysis. Since SIP is the control plane of VoIP, this is a big one!!
Session Initiation Protocol (SIP) provides the signaling used to establish, modify, and terminate many VoIP calls. When a call fails to complete, rings incorrectly, disconnects unexpectedly, experiences registration problems, or behaves differently than expected, the SIP message exchange often provides the best starting point for determining what happened and where the failure occurred.
A dedicated Wireshark SIP Profile helps engineers and technicians focus on the signaling information that matters during VoIP troubleshooting. By emphasizing SIP requests and responses, methods such as INVITE, REGISTER, ACK, CANCEL, and BYE, response codes, Call-ID, CSeq, addressing information, and other important SIP headers, the profile makes it easier to follow call setup and teardown and identify signaling errors or unexpected behavior.
For VoIP engineers and technicians, the profile provides a faster and more consistent way to investigate call signaling. Instead of repeatedly configuring Wireshark columns, filters, and display settings, the SIP Profile creates a purpose-built workspace for tracing call flows, examining registration activity, identifying failed transactions, and correlating signaling events with RTP, RTCP, and other packet-level evidence.