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<title>packeteer/tests/test_rtcp.cpp, branch main</title>
<subtitle>Packet capture and analysis, TUI and desktop GUI.
</subtitle>
<id>https://srdusr.com/git/packeteer/atom?h=main</id>
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<updated>2025-11-19T19:18:00+00:00</updated>
<entry>
<title>Add RTP/RTCP as a labeled heuristic fallback for unmatched UDP traffic</title>
<updated>2025-11-19T19:18:00+00:00</updated>
<author>
<name>srdusr</name>
<email>99972264+srdusr@users.noreply.github.com</email>
</author>
<published>2025-11-19T19:18:00+00:00</published>
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Architecturally different from every other protocol added so far: RTP
has no fixed well-known port at all - it's negotiated per call via
SDP/SIP/WebRTC signaling this project doesn't parse - so L7Registry's
port-keyed dispatch doesn't apply. Handled instead as a fallback tried
only when a UDP packet's normal port-based lookup finds nothing, with
every match labeled "?" (e.g. "RTCP? SR") to mark it as inferred from
packet shape rather than certain - the same honesty Wireshark itself
applies to heuristic dissection, which is off by default there for
exactly this reason.

The two heuristics aren't equally trusted, and the code says so: RTCP
checks a narrow packet-type range (200-204) plus an exact self-declared
length, both unlikely to occur by chance; RTP leans mostly on the 2-bit
version field, since its other structural checks are trivially
satisfied whenever those bits happen to be zero, the common case even
for unrelated traffic. Shipped anyway - a labeled guess on real
RTP/RTCP traffic is more useful than silence - but this is the first
place in the project where a match doesn't mean certainty.

Live-verified against genuine media traffic: ffmpeg streaming a real
RTP video test pattern to loopback, correctly decoded with incrementing
sequence numbers and a consistent SSRC across the stream, plus a real
RTCP Sender Report ffmpeg sent alongside it.
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