Wireshark VoIP: Capture, Analyze, and Fix VoIP Call Issues

When your VoIP calls drop, echo, or sound like they’re underwater, the problem isn’t always your headset or internet speed. It’s often hidden in the network traffic — and that’s where Wireshark VoIP, a free network protocol analyzer used to inspect real-time voice traffic. Also known as SIP packet sniffer, it lets you see exactly what’s happening between your phone and the server — down to the millisecond. Most people think VoIP issues are about bandwidth, but the real culprits are often misconfigured routers, firewall blocks, or jitter caused by competing devices on your network. Wireshark shows you the truth.

It doesn’t just capture data — it decodes SIP, the signaling protocol that sets up and ends VoIP calls, and RTP, the real-time protocol that carries the actual voice data. You’ll spot failed registrations, mismatched codecs, delayed packets, and even spoofed caller IDs. For example, if inbound audio is missing on recordings, Wireshark can show you if the RTP stream is being routed to the wrong port. If calls drop after 30 seconds, it might be a NAT timeout — visible in the SIP INVITE and 200 OK exchange. You don’t need to guess anymore.

Businesses use Wireshark VoIP to prove their ISP is throttling voice traffic. IT teams use it to validate QoS settings before rolling out new phones. Remote workers check their home networks for interference from streaming or gaming. And when your provider says "everything looks fine," Wireshark gives you the evidence to push back. It’s not magic — it’s just raw data made readable.

Below, you’ll find real-world guides on how to filter SIP traffic, identify RTP stream issues, decode SRTP encryption, and fix one-way audio using packet captures. No theory. No fluff. Just what works when your calls are failing.