Source note · Analysing a SIP call in Wireshark
For SRTP media captured without keys, sequence-number based loss and timing statistics can still be derived from the clear RTP header, but codec-dependent statistics and audio playback are not available.
inferred · 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
- Status
- inferred
- Checked
- 2026-09-24
Sources
- RFC 3711: The Secure Real-time Transport Protocol (SRTP) — IETF / RFC Editor · tier 1 standards and regulators · RFC 3711 section 3.1, combined with Wireshark User's Guide RTP Stream Analysis note on codec-dependent columns
Cite this note
APA
WarmTransfer. (2026, September 24). Analysing a SIP call in Wireshark: source note sip-call-analysis-wireshark-srtp-header-stats-inf. WarmTransfer. https://warmtransfer.net/knowledge/claims/sip-call-analysis-wireshark-srtp-header-stats-inf
BibTeX
@misc{warmtransfer-claim-sip-call-analysis-wireshark-srtp-header-stats-inf,
title = {Analysing a SIP call in Wireshark: source note sip-call-analysis-wireshark-srtp-header-stats-inf},
author = {{WarmTransfer}},
year = {2026},
url = {https://warmtransfer.net/knowledge/claims/sip-call-analysis-wireshark-srtp-header-stats-inf},
note = {Source note sip-call-analysis-wireshark-srtp-header-stats-inf, checked 2026-09-24}
}