Source note · Analysing a SIP call in Wireshark
The RTP Stream Analysis side panel summarises each stream's maximum delta, jitter, packet count, lost packets calculated from sequence numbers, duration and clock drift.
Checked 2026-09-24
- Vendor
- Wireshark
- Product
- Wireshark
- Subsystem
- RTP analysis
- Deployment
- on-premises
- Region
- not restricted
- Release range
- current as of 2026-09-24
- Checked
- 2026-09-24
Sources
- Wireshark User's Guide 9.11 RTP (RTP Streams, RTP Stream Analysis, RTP Player windows) — Wireshark Foundation · tier 2 current vendor documentation · RTP > RTP Stream Analysis Window (side panel statistics)
Cite this note
APA
WarmTransfer. (2026, September 24). Analysing a SIP call in Wireshark: source note sip-call-analysis-wireshark-rtp-analysis-summary. WarmTransfer. https://warmtransfer.net/knowledge/claims/sip-call-analysis-wireshark-rtp-analysis-summary
BibTeX
@misc{warmtransfer-claim-sip-call-analysis-wireshark-rtp-analysis-summary,
title = {Analysing a SIP call in Wireshark: source note sip-call-analysis-wireshark-rtp-analysis-summary},
author = {{WarmTransfer}},
year = {2026},
url = {https://warmtransfer.net/knowledge/claims/sip-call-analysis-wireshark-rtp-analysis-summary},
note = {Source note sip-call-analysis-wireshark-rtp-analysis-summary, checked 2026-09-24}
}