Keys pressed on the other phone reached imsd-media as RFC 4733 events and
were dropped with every packet that is not the audio's. The playout now
takes them: the gap before an event's start (a stock sender's DTX silence
often precedes a key press) is filled as before an audio frame, each
packet reports the event's duration so far from its start stamp, and what
the clock has not covered yet is synthesized as the event's Q.23
frequency pair at the packet's volume, so a retransmitted end packet adds
nothing, a lost update is caught up by the next, and a far end whose
audio resumes past the event (the stock phone's does) meets the clock
with the usual fill. An answer may renumber telephone-event from what we
offered (RFC 3264: the answerer sends with the offer's numbers, §6.1,
and the offerer with the answer's, §5.1), so the leg gets both: DTMF_PT
to send with, DTMF_RX_PT to take events at as well — our own SDP's
number: the offer's 98 or 100 on a call we placed, the offer's number we
echoed on one we answered; when the answer gave the audio our offer's
number, that number is the leg's audio and DTMF_RX_PT is the answer's
telephone-event number instead — and takes events at either number,
never at the audio's. --selftest pins the frequency table; the stats
gain rx_dtmf, tones and dtmf_rx_pt.
Tested: --selftest, tests/Sdp and tests/Engine (the eight suites pass);
the loopback's --dtmf-rx arm on a host build of this commit's tree
(journal/ims/captures/2026-10-15-dtmf-loopback-c3-rx.txt; the tx, both,
plain and collision runs beside it); the dev phone's dtmf benches ran in
builds of this series.
SendDtmf was a stub: GNOME Calls' SendTones and Plasma Dialer's sendDtmf
reached imsd and went nowhere. The SDP parser reads the telephone-event
payload type at the chosen codec's clock into the answer, among the
m-line's formats only — an rtpmap the line does not list is not
negotiated, and nothing is sent with a format the answer does not carry
(RFC 3264 §8.3.2); an event's rate is its rtpmap clock (RFC 4733
§2.4.1), an offer with two codec clocks lists telephone-event at each,
and the name is matched as every encoding name is, case-insensitively,
with the clock whole. The media leg carries it, and imsd-media gets it
as DTMF_PT with its stdin on a pipe from imsd, the control pipe. The
SDP answer imsd sends to an inbound offer carries telephone-event by the
same rule, where it echoed any telephone-event rtpmap with a lower-case
name and a clock prefix before: an rtpmap off the offer's m-line is no
longer echoed, an upper-case name now is. The decision is the engine's:
OnDtmf lets the tones through on an active call that negotiated
telephone-event (the answer's number on a call we placed, the offer's on
one we answered) and drops them otherwise with a line that says why:
before the answer, after the call, no telephone-event at the codec's
clock in the SDP, a character that is no tone or more than 128 tones
(then the whole string, never part of what was pressed); letters are
upper-cased. The shell writes one `dtmf <tones>` line per SendDtmf,
whole or not at all: the bound keeps the line within one atomic pipe
write, the write end is non-blocking so that a leg that stopped reading
fails the write instead of holding the engine thread, and a longer line
is refused; SIGPIPE is ignored so that a write after the leg died fails
with EPIPE instead of ending the daemon. The command queue's one
argument is `arg` now — the number to dial, or the tones. The `media …`
start line gains the telephone-event pt at its end.
Tested: tests/Sdp and tests/Engine (the eight suites pass); the dev
phone's dtmf benches ran in builds of this series
(journal/ims/captures/2026-10-14-dtmf-bench-1/ onward).
KPN's interconnect gateway transcodes every caller up: a G.711-only
fixed-line INVITE reached the phone offering PCMA, PCMU, AMR and AMR-WB
(bench call 2026-09-08), so the AMR-WB-only build rang and the narrowband
path — the one the Telia field report hit with 488 — cannot be reached on
the air through KPN by any caller.
CODECS=<list> (comma-separated over AMR-WB, AMR/AMR-NB, PCMA, PCMU) makes
the list both the codecs offered on an outgoing call and the acceptance
preference for an inbound offer, in that order: CODECS=PCMA takes G.711
A-law out of the mixed offer above, CODECS=PCMA,PCMU sends a G.711-only
offer toward the network. Unset, nothing changes — the default offer
bytes and the AMR-WB > AMR > PCMA > PCMU preference stay pinned. The
daemon logs an active override at startup.
Our offer listed AMR-WB alone. A call we place TO a landline then depends
on the far side transcoding up to wideband; a gateway that will not is
entitled to refuse. AMR narrowband is the 3GPP-mandatory codec every IMS
core and PSTN gateway can answer, so it goes second, with telephone-event
at the 8 kHz clock beside the existing 16 kHz one. AMR-WB stays first, so
a mobile-to-mobile call negotiates exactly what it did before; if the
network does answer AMR, the engine already starts the media leg on the
answered codec.
Byte pins in tests/Sdp and tests/Messages re-pinned (INVITE body 323 ->
466 bytes).
NOTE FOR RELEASE: this changes the MO INVITE on the path the arranged 112
calls used. One bench MO call on KPN before it ships.
An incoming INVITE whose offer carried no AMR-WB was refused with 488
before ringing. A landline caller arrives through the PSTN gateway, which
offers narrowband — AMR (NB) and/or G.711 — so every landline call was
refused, deterministically. Field report 2026-09-01 (Telia Norge): a
doctor's office called twice, both 488; a mobile caller rang fine.
The parser now recognises PCMA/PCMU by rtpmap name or by static payload
type (a G.711 offer may carry no rtpmap line at all, RFC 3551 §6), matches
encoding names case-insensitively (RFC 4566 §6), and prefers
AMR-WB > AMR > PCMA > PCMU with octet-aligned first within AMR — a mobile
caller offering everything still lands on AMR-WB. The engine accepts any
codec the media leg plays and answers at the offer's own payload type with
the DTMF clock matching the codec (8 kHz for narrowband). A refused offer
now logs what WAS offered ('offered: 18 G729, 101 TELEPHONE-EVENT'), so a
field journal answers 'which codec did the gateway want' without a raw SIP
dump.
The media leg gains the matching codecs in the next commit.
Userspace VoLTE/IMS daemon for mainline Linux phones (developed on the
Fairphone 6): a GLib-free core (SIP, SDP, USIM AKA, IPsec SA setup, RTP
media, call engine) behind a GDBus control daemon, a standalone AMR-WB
media leg, and a Plasma Dialer backend. C++26 modules built with Crafter
Build; the tree is clean under the project's house-style linter
(crafter-build lint) across all three build products.
Assisted-by: Claude:claude-fable-5
Signed-off-by: Jorijn van der Graaf <jorijnvdgraaf@catcrafts.net>
