prefixkit RU

VoIP & telephony

SIP message and log analyzer

Paste SIP messages or an Asterisk, FreeSWITCH, Kamailio or sngrep log to get a message ladder, SDP and codec analysis, NAT, DTMF and SRTP checks, and STIR/SHAKEN decoding.

Works with: pjsip set logger on, sip set debug on, sofia loglevel, sngrep (save as .txt), Wireshark “Follow → UDP Stream”, or a bare SDP. Everything runs in your browser; the text is not sent anywhere.

When to use it

When you have no capture but you do have the PBX log. Most problems are visible in signalling: a private IP in SDP (no audio), a missing ACK (call drops after 32 seconds), codec mismatch (488), a wrong password (repeated 401), DTMF not negotiated (menu keys do not work). If you have a pcap, load it into the pcap analyzer: it also measures audio quality.

STIR/SHAKEN

Carriers in the US and Canada sign the caller’s number: the Identity header carries a PASSporT token with attestation A, B or C. A means the carrier knows the customer and their right to the number. Calls with C or no signature are more often flagged as spam.