Distributed SIP capture
Start a capture by caller, called number, tenant code, SIP domain, provider IP, free text, or known Call-ID after a call already exists.
Telephony Unified Network Analyzer helps operators trace calls across edge SBCs, access switches, PBX SBCs, and Asterisk or FreeSWITCH nodes from one coordinated interface.
TunaIO coordinates agents on every relevant node, starts bounded captures with explicit search criteria, and streams matching signaling and log context back to one command center.
Use TunaIO when packet capture, SIP ladder review, RTP inspection, and application logs all need to line up in one timeline.
Start a capture by caller, called number, tenant code, SIP domain, provider IP, free text, or known Call-ID after a call already exists.
Optionally collect RTP streams for timing and quality analysis when one-way audio, dropouts, or media failover behavior needs proof.
Pull matching Kamailio, FreeSWITCH, and Asterisk log lines into the same session so the route decision and SIP trace are visible together.
Run TunaIO as a managed command center, deploy it inside your network, or use a hybrid model where agents report to your selected analysis server.
Agents run near your SIP, RTP, and application logs on SBCs, PBXs, and media gateways.
Every capture requires at least one explicit search criterion and expires automatically.
See call summaries, expanded ladders, raw SIP payloads, and correlated logs from the same session.