Packet captures for VoIP troubleshooting
Use a small, repeatable packet-capture method to diagnose SIP failures, missing RTP, DTMF problems, and call-quality symptoms.
From the blog
Clear explanations of VoIP, SIP, RTP, codecs, phone numbers and the technologies behind modern voice.
Use a small, repeatable packet-capture method to diagnose SIP failures, missing RTP, DTMF problems, and call-quality symptoms.
Learn what a SIP trunk is, how a PBX connects to a carrier, and which signalling, media, numbering, and security checks matter.
Plan RTP and RTCP port ranges, firewall rules, NAT behaviour, and concurrent call capacity without confusing signalling and media.
Understand in-band, RFC 2833 or RFC 4733, and SIP INFO DTMF methods, including when IVR keypad tones fail.
Learn how A, AAAA, NAPTR, and SRV records help phones and SIP systems find services, choose transports, and fail over.
Understand the roles of SIP, SDP, RTP, RTCP, TLS, SRTP, STUN, TURN, ICE, DNS, and related protocols in an IP call.
Compare common voice codecs by payload rate, speech bandwidth, interoperability, delay, and practical deployment considerations.
Understand common SIP status codes, what they mean, and what to check when registration or call setup fails.
Learn how delay, jitter, packet loss, and congestion affect voice calls, and use a practical checklist to improve call quality.
Map common VoIP symptoms and evidence to the physical, data-link, network, transport, session, presentation, and application layers.
Diagnose registration failures, one-way audio, and dropped calls caused by NAT, firewalls, and incorrect SIP or RTP address handling.
Use a compact reference for IP addressing, subnets, routing, DNS, ports, NAT, and network checks relevant to VoIP.