diff --git a/AGENTS.md b/AGENTS.md index 7cf5274..a01b265 100644 --- a/AGENTS.md +++ b/AGENTS.md @@ -40,7 +40,7 @@ PipeWire. The verified device/carrier combinations are in the README. | `interfaces/Imsd.cppm` | The primary module interface of `imsd-core`: `export import`s the eight partitions below. | | `interfaces/Imsd-Util.cppm` | `imsd::util` — SIP tokens, IPv6 host/port formatting, the RFC 5952 §4 IPv6 text (`CanonicalIp6`/`CanonicalAddr`; an IPv4-mapped address spelled in hex stays hex, where §5 and `inet_ntop` write it dotted) and the comma-separated address-list parser (`ParseAddressList`), hex/base64, MD5 (RFC 1321), the fixed protected ports `PortUc`/`PortUs` (45061/45062). | | `interfaces/Imsd-Sip.cppm` | `imsd::sip` — SIP as a text layer plus TCP framing: header extraction, status codes, the granted lifetime, `ResponseOf` transaction matching. Not a full RFC 3261 parser, deliberately. | -| `interfaces/Imsd-Sdp.cppm` | `imsd::sdp` — SDP offer/answer: the AMR-WB + AMR + telephone-event offer (optional IR.92 preconditions), the parser the media plane needs, `BuildAnswer` for the terminating side, the `CODECS` override. | +| `interfaces/Imsd-Sdp.cppm` | `imsd::sdp` — SDP offer/answer: the AMR-WB + AMR + telephone-event offer (optional IR.92 preconditions; `OfferedTelephoneEventPt` names its telephone-event numbers), the parser the media plane needs (the codec, its payload type and mode, and the telephone-event payload type the m-line lists at that codec's clock), `BuildAnswer` for the terminating side, the `CODECS` override. | | `interfaces/Imsd-Aka.cppm` | `imsd::aka` — USIM AKA glue: the AKAv1-MD5 digest (RFC 3310), IMPI/IMPU/realm from IMSI + MCC/MNC, parsers for `qmicli --uim-get-card-status` and `mmcli` output, the AUTHENTICATE APDU and its response. | | `interfaces/Imsd-Ipsec.cppm` | `imsd::ipsec` — the two ESP SA pairs (TS 33.203) as `ip xfrm` argv lists, the Security-Server mechanism selection, and the reader that reconstructs SPIs/ports from `ip xfrm state`/`policy` text for a warm resume, and (1.0.0) `SaPeer`, the P-CSCF the kernel's pair is to, read from the `dir out` policies that carry imsd's fingerprint, `OwnXfrmDeleteCommands`, the deletes by exact key of what two rules select — the policies carrying imsd's fingerprint and the reqid-1/2 transport-mode ESP states in their direction or on the UE's ims-PDN address, whoever installed them (§12), and `FileNamesAnotherPair`, whether a state file's `spi_uc`/`spi_us` name the kernel's pair (unknown on either side decides nothing). | | `interfaces/Imsd-Messages.cppm` | `imsd::msg` — every SIP request and response imsd sends, built from an explicit `Context` and per-request variables so the builders are pure functions; byte-pinned by `tests/Messages`. Also `IsEmergency`. | @@ -48,7 +48,7 @@ PipeWire. The verified device/carrier combinations are in the README. | `interfaces/Imsd-Qmi.cppm` | `imsd::qmi` — the QMI WDS request bytes `imsd-pcscf` sends (Bind Mux Data Port, Set IP Family, Get Packet Service Status, Get Current Settings, Get Profile Settings), the message/TLV parser, the P-CSCF list readers (0x22 flag, 0x23 IPv4, 0x24 FQDN, 0x2E IPv6 — the last one libqmi does not model). | | `interfaces/ImsdBus.cppm` | `imsd::bus` — the polkit gate in front of call control, shared by imsd and imsd-ofonod. GLib, so outside `imsd-core`. | | `implementations/main.cpp` | The `imsd` daemon shell: GLib main loop, GDBus service, sockets, subprocesses, the registration sequence and the run loop that pumps the core. | -| `implementations/media.cpp` | `imsd-media`, the per-call RTP leg. Pure std + POSIX, dlopen's the AMR codecs. | +| `implementations/media.cpp` | `imsd-media`, the per-call RTP leg, DTMF included (RFC 4733 events out, queued by `dtmf ` lines on its stdin; the far end's events in, played as tones). Pure std + POSIX, dlopen's the AMR codecs. | | `implementations/pcscf.cpp` | `imsd-pcscf`, the P-CSCF reader: the AF_QIPCRTR socket, the WDS service lookup and the Bind Mux Data Port / Set IP Family / Get Current Settings sequence around `Imsd:Qmi` (plus Get Profile Settings with `--profile`, Get Packet Service Status with `-v`). std + POSIX + the Linux QRTR UAPI (``), no GLib; never Start/Stop Network. | | `implementations/dialerd.cpp` | `imsd-dialerd`, the Plasma Mobile session-bus backend (`org.kde.telephony.*`). | | `implementations/ofonod.cpp` | `imsd-ofonod`, the GNOME Calls system-bus backend (`org.ofono`). | @@ -282,13 +282,20 @@ What the suites pin: keepalive discards), `ResponseOf`, refused finals with reason headers, reginfo documents. A change that flips one of those tests is a behaviour change, not a cleanup. -- `Sdp` — the offer's exact bytes, the parser, `BuildAnswer` at the offer's +- `Sdp` — the offer's exact bytes, the parser (the telephone-event payload + type at the chosen codec's clock among it), `BuildAnswer` at the offer's own payload types and octet-align mode, the `CODECS` override. - `Engine` — the call state machine driven through recorded network dialogs (reliable 18x/PRACK, 2xx ACK + media start, far-end hangup, CANCEL/answer races, busy/error finals, the terminating side's 100/180/200/486/487/488, G.711-only and AMR-NB gateway offers, ring timeout, BYE both ways, the - emergency chain) and `RegRecovery` (thresholds, backoffs, the throttle + emergency chain, DTMF — sent to the leg on an active call that negotiated + telephone-event, dropped with the reason before the answer, after the + call, without the negotiation, for a character that is no tone or for a + string past `DtmfMaxTones`; the receive payload type: our offer's on a + renumbered answer, the answer's when the audio took ours, the echoed + offer's on an incoming call) and + `RegRecovery` (thresholds, backoffs, the throttle schedule, Retry-After, which waits a Dial may cut short, when the old registration counts as gone, the holds a restart takes over, a binding the network terminated). @@ -427,7 +434,57 @@ Rules: the script's keys ran on the host build (2026-10-12, exit 0), its own idle (nothing to register to), the discovery-failed exit 1, `main()`'s removal of a pair the holds wrote off and `imsd-ofonod` starting without waiting - for imsd on the dev phone (2026-10-12). + for imsd on the dev phone (2026-10-12). DTMF ran on the dev phone (KPN, + 2026-10-14 to 16, the fp6 journal's `2026-10-14-dtmf-bench-1` to + `2026-10-16-dtmf-bench-17-heard`): outgoing calls to voicemail over the bus + (AMR-WB, telephone-event 98 — three tones, a string with a character + that is no tone, a wrong call uni: the two drops' lines; benches 1, 5 + to 9 and 11 to 16), 13 keys from GNOME Calls' keypad on a call to a stock phone the + network answered narrowband (AMR 99, telephone-event 100), heard there + (bench 2, the first sender; 28 keys from Plasma Dialer's keypad on two + such calls heard there on the one-sender shape, bench 17); on incoming + calls from that phone (the offer's 111 at 16 kHz) + 10 and 4 keys sent from the keypad (benches 3 and 4); on Plasma Mobile, + 11 keys from Plasma Dialer's keypad on an incoming call and 3 on an + outgoing narrowband one, through `imsd-dialerd` (bench 10); the stock + phone's key presses heard as tones (on GNOME Calls: bench 3's twelve on + the renderer before the gap fill, bench 4's four on it with the gap + fill; on Plasma Mobile: bench 10's ten on an incoming wideband call and + two on an outgoing narrowband one, every received event rendered by the + leg's own count). Every event of + ours in benches 1 to 3 and 5 to 17 is in the RFC 4733 shape on the wire + with the clock shared with the audio, and so are three of bench 4's + four: the third carried a mic frame after its first packet (seq 1761 + between 1760 and 1762) — the race of the sender's first shape, a thread + of its own beside the mic's, which the sender's commit closes by having + the mic thread send the event packets in its own frames' slots (one + event of bench 4's four raced, none of the 23 live-mic events of benches + 2 and 3; the loopback's mic is a file, paced like a microphone, and has + not raced it). That first shape also stamped the audio a slot early or + late per event against the capture clock on the AMR-WB calls measured — + there pw-record delivers frames in bursts of two (§13) and the thread's + 140 ms window took six to eight of them: the two tones of `2#`, 10 slots + apart by construction, are 11, 9 and 10 slots apart in benches 5, 12 and + 13, AMR-WB voicemail calls (`analysis-v2-bench-{5,12,13}.txt` beside + bench 14) — which the one-sender shape of bench 14 cannot (10); on a + narrowband call the first shape's stamping was not measured. + Not reached on hardware: + a network answering without telephone-event (KPN echoes it; an answer that + keeps the rtpmap off its m-line is pinned in `tests/Sdp`), the drops + before the answer and after the call and the `DtmfMaxTones` bound + (these three pinned in `tests/Engine`), the letters A–D and lower case + (pinned in `tests/Engine` and the selftest), an inbound offer without + telephone-event at the codec's clock (pinned in `tests/Sdp`), the shell's + refusal of a line over PIPE_BUF and its `not sent: no control pipe` line + on a call with no media leg (nothing pins these two), a `pipe()` failure, a write to a + dead leg's pipe (`DTMF '…' not sent: …`), a full pipe's EAGAIN, G.711 + (the loopback only), a received event with a code past D (dropped + unrendered) or one whose packet reports more than two seconds the clock + has not covered (that packet skipped — a long key press renders slot by + slot), a lost first packet of an event (the renderer starts where the + clock stands), and an answer that renumbers telephone-event or gives the + audio our number (`DTMF_RX_PT` — KPN echoes our numbers; pinned in + `tests/Engine` and by the loopback's collision arm). - `imsd-pcscf` ran on the dev phone (KPN) on 2026-10-11 against the live ims bearer: the `--mux 1 --profile 2 -v` sequence, a plain run, and `OutOfCall` on an unbound mux (the fp6 journal's `2026-10-11-pcscf-pco-probe` capture, @@ -439,9 +496,23 @@ Rules: QRTR control lookup and the matching of answers by type, transaction and message id are shell code (`pcscf.cpp`) no test reaches. - `imsd-media --selftest` round-trips the RFC 4867 payload packing (both - formats, both AMR codecs) and the G.711 tables. `MIC_SRC=` (raw s16 + formats, both AMR codecs), the G.711 tables, and pins the RFC 4733 event + payload bytes, the keypad-to-event codes and every event's Q.23 + frequency pair. `MIC_SRC=` (raw s16 PCM at the codec rate, paced in real time) and `PCM_DUMP=1` are test seams - for driving the leg against a synthetic RTP peer without a phone. + for driving the leg against a synthetic RTP peer without a phone; the fp6 + journal's `utilities/media_loopback.py` is that peer, and its `--dtmf` + arm writes a `dtmf ` line to the leg's stdin and checks the events + on the wire (one timestamp per event, the marker, the growing duration, + the end bit three times, the shared sequence and timestamp space with the + audio; with the mic file, or without a mic — `--no-mic`, the silence tick + carrying the events), and its `--dtmf-rx` arm has the peer send events and checks the + decoded downlink for each tone's two frequencies where the event sat, the + stream intact around it — the sender's shape is proven there and on the + dev phone's wire (the fp6 journal's dtmf benches: their pcaps go through + the same checks), the renderer's there — and on the dev phone by ear and + by the leg's count of events rendered (bench 10); what a gateway does + with the events is proven only by a call to one. - `imsd --session` and `imsd-ofonod --session` own their names on the session bus for development without root, polkit or a phone (§8.3). @@ -504,8 +575,9 @@ the status snapshot live (main thread only). Per answered call imsd forks `imsd-media` (`IMSD_MEDIA`, default next to the `imsd` binary, else `/usr/libexec/imsd-media`) with the addresses, ports and -payload type as arguments and the codec and audio knobs in its environment -(§7.6, §10.2). +payload type as arguments, the codec and audio knobs in its environment +(§7.6, §10.2) and its stdin on a pipe from imsd, the control pipe the DTMF +tones travel down (§7.6). The desktop daemons are translation layers: `imsd-dialerd` runs as the session user (autostarted from `/etc/xdg/autostart/imsd-dialerd.desktop` in @@ -993,8 +1065,8 @@ journal says it was not handed over. `CallMachine` owns one call's SIP dialog, outgoing or incoming, and never touches a socket, a clock or a subprocess: each event returns `Action`s for the shell (send this on the client flow, answer that request on the channel -it came in on, start/stop the media leg, emit this D-Bus state change, arm -this deadline). States and reasons are the frozen D-Bus vocabulary: +it came in on, start/stop the media leg, hand it the tones a key press +sent, emit this D-Bus state change, arm this deadline). States and reasons are the frozen D-Bus vocabulary: `dialing` / `ringing` / `incoming` / `active` / `terminated`, with reasons `outgoing` / `incoming` / `accepted` / `local-hangup` / `remote-hangup` / `refused-or-busy` / `error`. @@ -1007,7 +1079,8 @@ this deadline). States and reasons are the frozen D-Bus vocabulary: - Incoming: an INVITE on a protected-port listener is validated before it is announced to the UI; 100 and 180 (reliable when the caller requires 100rel); `Accept` sends the 200 with the SDP answer at the offer's own - payload types and octet-align mode; a remote CANCEL gets 200 + 487; an + payload types and octet-align mode, its telephone-event echoed when the + offer's m-line lists one at the codec's clock; a remote CANCEL gets 200 + 487; an unusable offer 488 naming what was offered; ring timeout 480. A landline gateway's G.711-only or AMR-NB offer rings and is answered at its own payload types. @@ -1016,7 +1089,26 @@ this deadline). States and reasons are the frozen D-Bus vocabulary: `urn:service:sos` first, with one fallback to a plain INVITE of the digits; a plain INVITE answered `380 Alternative Service` with an emergency body is re-attempted as `urn:service:sos`. -- `SendDtmf` is a stub (RFC 4733 not implemented). +- DTMF (`SendDtmf`): the tones go out as RFC 4733 telephone-event packets + on the call's RTP stream, at the payload type the SDP negotiated for the + codec's clock (`MediaLeg::dtmfPt`: the answer's for an outgoing call, the + offer's for an incoming one; the leg takes the far end's events at that + number and at `MediaLeg::dtmfRxPt`, `DTMF_RX_PT` — our own SDP's at that + clock: our offer's on a call we placed (the answerer sends with the + offer's numbers, RFC 3264 §6.1), the echoed offer's on one we answered + (the offerer with the answer's, §5.1); the answer's instead when + the answer gave the audio our offer's number, which is then the leg's + audio number — never at the audio's). The + engine's `OnDtmf` lets them through on an active call that negotiated + telephone-event and drops them otherwise, with a journal 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 a string past + `DtmfMaxTones` (128) — then + the whole string, never part of what was pressed; letters are + upper-cased, `a` and `A` are one event). There + is no second method: no SIP INFO, and no tone played into the microphone + path. The shell writes one `dtmf + ` line per `SendDtmf` down the leg's control pipe. - Incoming calls reach imsd only when the network's T-ADS routes the call to the IMS domain; the REGISTER Contact advertises the MMTEL ICSI and `+sip.instance` for that. When T-ADS still picks CS, no INVITE arrives — @@ -1032,7 +1124,35 @@ narrowband and G.711 PCMA/PCMU (8 kHz, what a PSTN gateway offers); AMR frames ride RFC 4867 payloads, octet-aligned or bandwidth-efficient (`OCTET_ALIGN`, mirrored from the SDP). The AMR codecs are dlopen'd (`libvo-amrwbenc` + `libopencore-amrwb` for WB, `libopencore-amrnb` for NB); -G.711 is a table. **Exit code 3 means the downlink dried up** for +G.711 is a table. DTMF (RFC 4733 telephone-events; `DTMF_PT` names their +payload type, unset = none; imsd sets it from the SDP and writes the lines +down the control pipe, §7.1): each `dtmf +` line on its stdin queues the tones, and a tone takes seven 20 ms +slots of the uplink in place of the audio frames — the event's five +(100 ms at -10 dBm0: the marker bit on the first packet, the duration +growing a slot at a time, the end bit on the fifth packet) and two in +which the end packet goes out again (RFC 4733 §2.5.1.4) — then three +slots of audio before the next tone; the audio's own sender carries the +events — per captured frame (a silence tick without a mic) the event +packet goes in the frame's place, one packet per slot — so events and +audio share the stream's sequence and timestamp space (RFC 4733 §2.1), the +clock stays the capture's, and the audio after an event is stamped as if +it had run through it; a line of more than 4096 bytes is +dropped through its end and the next line goes. The far end's events — +taken at `DTMF_RX_PT` (our offer's 98 or 100 on a call we placed, the +number an answerer sends with, RFC 3264 §6.1, whatever the answer +renumbered ours to; the offer's number we echoed on a call we answered; +the answer's telephone-event number instead when the answer gave the +audio our offer's number, which is then the leg's audio number) and at +`DTMF_PT`, never at the audio's — are played as the tones they stand for +(ITU-T Q.23's frequency pairs, at the packet's volume), in the playout, in +step with the audio clock: the gap before the event's start (a stock +sender's DTX silence often precedes a key press — 0 to 7 slots on the dev +phone, 2026-10-15) is filled as before an audio frame, each packet reports +the event's duration so far and what the clock has not covered is +synthesized, so a retransmitted end packet adds nothing and a lost update +is caught up by the next; a far end whose audio resumes past the event +(the stock phone's does, §13) meets the clock with the usual fill. **Exit code 3 means the downlink dried up** for `MEDIA_TIMEOUT` seconds (default 6.0) — the far-end-hangup signal, which on carriers whose BYE never reaches our SAs is the reliable teardown trigger. Per-call stats land in `OUT_DIR` (default the state dir); `RTP_DUMP=1` @@ -1127,6 +1247,20 @@ Lines the operator (and the scripts) look for, all prefixed `imsd: `: failed: … — the resume is given up (the engine's rule): the pair is removed`). - `audio session: …` at start and at each call's media start; `CODECS override active: …`. +- `media pt=N octet=true|false -> [X]:P, telephone-event pt N|none` at a + call's media start (`… pt N (theirs M)` when `DTMF_RX_PT` differs from + the one we send with); `call X: DTMF '…' -> media leg (telephone-event pt + N)` for tones the engine lets through, `call X: DTMF '…' dropped: the call is ` / + `… dropped: no telephone-event negotiated for this call` / `… dropped: + 'c' is not a tone` / `call X: DTMF of N tones dropped: more than 128` for + tones not sent, `SendDtmf(X, '…') ignored: no such call`; `DTMF '…' not + sent: no control pipe to the media leg` on a call whose control pipe + could not be made or that has no media leg, `DTMF '…' not sent: ` + when the control pipe's write fails (`EPIPE`: the leg died under the + action; `EAGAIN`: it is alive, stopped reading, its pipe full), + `DTMF of N tones not sent: longer than the control pipe's atomic + write` (a line the engine's bound should have stopped), `media control + pipe: …; this call has no DTMF` when the pipe could not be made. - A refused REGISTER or SUBSCRIBE is logged with its status line and any `Retry-After` / `Warning` / `Reason` / `Server` headers; `re-REGISTER challenged: 401 …; re-authentication asked, the next @@ -1257,8 +1391,6 @@ Known gaps, all inherited rather than introduced: `mm_sim_dup_emergency_numbers()`). An upstream fix would close this. Dialling an emergency number from the dialpad is unaffected — that reaches imsd's own classifier. -- DTMF is forwarded to imsd, where `SendDtmf` is a stub — the same - limitation Plasma has. - Restarting `imsd-ofonod` restarts GNOME Calls with it: when the modem's `VoiceCallManager` interface goes away and comes back, Calls 50.0 adds a new origin but leaves the old origin's `CallAdded` handler connected, so @@ -1398,12 +1530,12 @@ resume. ### 10.2 imsd-media (per call, set through imsd's environment) -The README's second table. Two pairs are not configuration and are left out +The README's second table. These are not configuration and are left out there: | Variable | Default | Meaning | |---|---|---| -| `CODEC`, `OCTET_ALIGN` | *(from the SDP)* | set by imsd from the negotiated answer; not configuration | +| `CODEC`, `OCTET_ALIGN`, `DTMF_PT`, `DTMF_RX_PT` | *(from the SDP)* | set by imsd from the SDP (`DTMF_PT`: the telephone-event payload type at the codec's clock the leg sends with, empty when none was negotiated; `DTMF_RX_PT`: a second number the leg takes events at — our own SDP's at that clock, or the answer's when the answer gave the audio our offer's number, §7.6); not configuration | | `MIC_SRC`, `PCM_DUMP` | *(unset)* | test seams (§5) | ### 10.3 ims-pdn-up.sh and imsd-ofonod @@ -1543,6 +1675,40 @@ the related code. - **Modem attach-family forcing** (Odido, O2 UK, Swisscom): an APN-string IPv6 bearer request is refused when the attach PDN was granted IPv4-only; a profile-indexed request is not. +- **pw-record delivers 20 ms frames in bursts** — of two on an AMR-WB + call (1.8 frames per burst on average, the bursts ~37 ms apart; the dev + phone, 2026-10-15/16: of the uplink's inter-packet gaps 42–45 % are + under 4 ms and 50–52 % are 30–40 ms, 3–8 % lie between and under 1 % + above — the fp6 journal's `2026-10-16-dtmf-bench-14/uplink-cadence.txt` + over benches 4, 5, 13 and 14, counted by the `cadence.py` beside it), + of three or four on a narrowband one (3.3 on average, ~68 ms apart; + 70 % under 4 ms, 28 % over 40 ms — `2026-10-16-dtmf-bench-17-heard/uplink-cadence-narrowband.txt`, + bench 17's two calls; the burst sizes of all six legs in `burst-sizes.txt` + beside it). A sender on a clock of its own cannot count the frames a + 140 ms window holds, so the events ride the frames' own slots (§7.6), + and an event's packets leave in those bursts: the audio's own spacing, + which RFC 4733 §2.5.1.2 calls the natural interval for event updates + (its 50 ms recommendation is for a spacing of the sender's own). +- **DTMF on KPN (2026-10-14 to 16):** the answer to our offer echoes + telephone-event at our payload types — 98 at 16 kHz on an AMR-WB call to + voicemail, 100 at 8 kHz on a phone-to-phone call the network answered + narrowband AMR at 99 — and a stock phone's terminating INVITE offers + AMR-WB 107 with telephone-event 111; events of the sender's first shape + (100 ms, a packet every 20 ms, at least 100 ms apart) were heard on that + phone (bench 2 the first shape; bench 17 the one-sender shape — 28 keys + on two narrowband calls, every one heard there). On + its own AMR-WB calls to us (telephone-event 111 at 16 kHz) that phone + sends one event of 20 packets per key press (340 ms, -10 dBm0, an update + every 20 ms, the end packet three times), its audio resuming one to eight + slots past the event's end and the event preceded by a DTX gap of up to + seven slots (the fp6 journal's benches 2 to 4 and bench 10's incoming + call, their `far-end-events-*` files); on a call we placed that the + network answered narrowband (telephone-event 100 at 8 kHz) its two key + presses arrived as events of 22 and 19 packets (400 and 340 ms), the + audio resuming 15.8 and 19 slots (316 and 380 ms) past their ends, the + first off the stream's 20 ms timestamp grid (bench 10's outgoing call) — + whether that shape is the phone's or the network's interworking, + nothing on disk says. - **pmOS's default nftables** drops inbound on `qmapmux*`; without the protected-port rule every network-initiated request (NOTIFY, MT INVITE) dies after ESP decapsulation and MT calls fall back to CS. diff --git a/README.md b/README.md index 826b4bb..b2f35de 100644 --- a/README.md +++ b/README.md @@ -6,28 +6,19 @@ Userspace IMS/VoLTE daemon for mainline Linux phones. Most Linux phones drive the modem's internal IMS engine. -imsd implements the whole IMS client in userspace instead: -SIP registration with USIM AKA authentication over kernel -IPsec (ESP), call signalling, and the AMR-WB RTP media plane wired into +imsd implements the whole IMS client in userspace instead wired into PipeWire so a stock mobile dialer like Plasma Dialer or GNOME Calls under Phosh can place and receive real VoLTE calls on a commercial network. -## Status - -Call-capable, both directions: outgoing calls place, answer, carry full-duplex -AMR-WB audio and tear down. Incoming calls ring, answer and carry media the -same way, driven end-to-end from a stock dialer over a frozen D-Bus ABI. -Not there yet: DTMF (`SendDtmf` is a stub), hold/multiparty/call waiting, -USSD. - -Verified by community members to be fully working: +Calls verified by community members to be fully working: | Device | OS | Carrier | |---|---|---| -| The Fairphone (Gen. 6) | postmarketOS, Linux 7.1.2 | KPN NL | -| The Fairphone (Gen. 6) | postmarketOS, Linux 7.1.2 | Telekom Deutschland GER | -| The Fairphone (Gen. 6) | postmarketOS, Linux 7.1.2 | Phonero | -| The Fairphone (Gen. 6) | postmarketOS, Linux 7.1.2 | Telia Norge | +| The Fairphone (Gen. 6) | Nura, Linux 7.1.2 | KPN NL | +| The Fairphone (Gen. 6) | Nura, Linux 7.1.2 | Telekom Deutschland GER | +| The Fairphone (Gen. 6) | Nura, Linux 7.1.2 | Phonero | +| The Fairphone (Gen. 6) | Nura, Linux 7.1.2 | Telia Norge | +| The Fairphone (Gen. 6) | Nura, Linux 7.2.0 | o2-uk | ## What it needs diff --git a/implementations/main.cpp b/implementations/main.cpp index 08c5c70..6af25b2 100644 --- a/implementations/main.cpp +++ b/implementations/main.cpp @@ -31,6 +31,7 @@ reach the bus via g_idle_add, where the calls table + status snapshot live #include #include #include +#include #include #include #include @@ -48,7 +49,7 @@ import ImsdBus; namespace { // ---- static config (env-overridable, same knobs as imsd.py) --------------- -constexpr const char* Version = "1.0.0"; +constexpr const char* Version = "1.1.0"; constexpr const char* BusName = "net.catcrafts.IMS1"; constexpr const char* ObjPath = "/net/catcrafts/IMS1"; constexpr const char* Iface = "net.catcrafts.IMS1"; @@ -1052,6 +1053,10 @@ public: std::scoped_lock g(cmdLock_); cmds_.push_back({Cmd::Kind::Accept, uni, {}}); } + void SendDtmf(const std::string& uni, const std::string& tones) { + std::scoped_lock g(cmdLock_); + cmds_.push_back({Cmd::Kind::Dtmf, uni, tones}); + } void Stop() { { std::scoped_lock g(cmdLock_); cmds_.push_back({Cmd::Kind::Stop, {}, {}}); } quit_.store(true); @@ -1140,7 +1145,7 @@ public: } private: - struct Cmd { enum class Kind { Dial, HangUp, Accept, Stop } kind; std::string uni, number; }; + struct Cmd { enum class Kind { Dial, HangUp, Accept, Dtmf, Stop } kind; std::string uni, arg; }; // arg: the number to dial, or the tones struct Stopping {}; // the engine was told to stop while a bring-up waited // A bring-up the network deferred (a fresh-SA throttle) or refused: wait @@ -1247,6 +1252,7 @@ private: bool dropCall_ = false; bool closeFlowWhenIdle_ = false; // a binding was terminated under a call: its flow closes with the call pid_t mediaPid_ = -1; + int mediaCtl_ = -1; // our end of the leg's control pipe (its stdin) double deadline_ = 0; // reply_ answers the inbound request being dispatched right now; // callReply_ is the call's durable response channel — refreshed from @@ -2368,8 +2374,8 @@ private: bool ClearForDial(const Cmd& c, bool emergency) { if (!call_) return true; if (!emergency) { - Log(std::format("dial({}) refused: call in progress", c.number)); - CallInfo bad{c.uni, c.number, "terminated", "error", NowEpoch(), 0}; + Log(std::format("dial({}) refused: call in progress", c.arg)); + CallInfo bad{c.uni, c.arg, "terminated", "error", NowEpoch(), 0}; PostAdded(bad); PostState(c.uni, "terminated", "error"); PostDeleted(c.uni); @@ -2385,7 +2391,7 @@ private: void HandleCmd(const Cmd& c) { if (c.kind == Cmd::Kind::Stop) { quit_.store(true); return; } if (c.kind == Cmd::Kind::Dial) { - bool emergency = imsd::msg::IsEmergency(c.number, emergencyExtra_); + bool emergency = imsd::msg::IsEmergency(c.arg, emergencyExtra_); if (!ClearForDial(c, emergency)) return; // a dead client flow fails the INVITE instantly — give the // recovery path one immediate shot first: user action cuts the @@ -2404,9 +2410,9 @@ private: if (emergency) Log(std::format("EMERGENCY dial ({}): urn:service:sos over the live registration, " "fallback plain INVITE — stage 1, no emergency registration/PDN; " - "carrier-side behavior UNVERIFIED", c.number)); - call_.emplace(ctx_, rng_, c.uni, c.number, rtpPort_, precond_, emergency, codecs_); - CallInfo info{c.uni, c.number, "dialing", "outgoing", NowEpoch(), 0}; + "carrier-side behavior UNVERIFIED", c.arg)); + call_.emplace(ctx_, rng_, c.uni, c.arg, rtpPort_, precond_, emergency, codecs_); + CallInfo info{c.uni, c.arg, "dialing", "outgoing", NowEpoch(), 0}; PostAdded(info); bool ok = Execute(call_->Start()); if (!ok && call_ && !call_->Terminated()) { Execute(call_->Fail("error")); } @@ -2416,6 +2422,9 @@ private: } else if (c.kind == Cmd::Kind::Accept) { if (call_ && call_->Uni() == c.uni) { Execute(call_->OnAccept()); MaybeDropCall(); } else Log(std::format("Accept({}) ignored: no such ringing call", c.uni)); + } else if (c.kind == Cmd::Kind::Dtmf) { + if (call_ && call_->Uni() == c.uni) Execute(call_->OnDtmf(c.arg)); + else Log(std::format("SendDtmf({}, '{}') ignored: no such call", c.uni, c.arg)); } } @@ -2515,6 +2524,7 @@ private: break; case T::StartMedia: StartMedia(a.media); break; case T::StopMedia: StopMedia(); break; + case T::SendDtmf: SendDtmfToMedia(a.text); break; case T::State: if (announce) PostState(call_->Uni(), a.state, a.reason); break; @@ -2541,7 +2551,9 @@ private: void StartMedia(const imsd::engine::MediaLeg& leg) { if (!leg.Valid()) { Log("SDP answer missing media endpoint; no media leg"); return; } - Log(std::format("media {} pt={} octet={} -> [{}]:{}", leg.codec, leg.payloadType, leg.octetAlign, leg.remoteIp, leg.remotePort)); + Log(std::format("media {} pt={} octet={} -> [{}]:{}, telephone-event pt {}{}", leg.codec, leg.payloadType, leg.octetAlign, leg.remoteIp, leg.remotePort, + leg.dtmfPt ? std::to_string(*leg.dtmfPt) : std::string("none"), + leg.dtmfRxPt && leg.dtmfRxPt != leg.dtmfPt ? std::format(" (theirs {})", *leg.dtmfRxPt) : std::string())); if (Is6(leg.remoteIp)) RunCmd({"ip", "-6", "route", "replace", leg.remoteIp, "dev", dev_}); else RunCmd({"ip", "route", "replace", leg.remoteIp, "dev", dev_}); @@ -2555,8 +2567,21 @@ private: session = ResolveAudioSession(); Log(AudioSessionLine(session) + (session.name.empty() ? "; this call has no audio" : "")); } + // the control pipe: imsd-media reads `dtmf ` lines on its + // stdin (§7.6); our end closes with the leg, no other child inherits + // it, and it never blocks the engine thread — a leg that stopped + // reading fails the write (EAGAIN once the pipe is full) instead + int ctl[2] = {-1, -1}; + if (pipe(ctl) != 0) { + Log(std::format("media control pipe: {}; this call has no DTMF", std::strerror(errno))); + } else { + fcntl(ctl[0], F_SETFD, FD_CLOEXEC); + fcntl(ctl[1], F_SETFD, FD_CLOEXEC); + fcntl(ctl[1], F_SETFL, O_NONBLOCK); + } pid_t pid = fork(); if (pid == 0) { + if (ctl[0] >= 0) dup2(ctl[0], STDIN_FILENO); // dup2 clears the close-on-exec if (geteuid() == 0) { // no session is said with an empty value: imsd-media reads a // missing variable as a caller older than 0.3.10 @@ -2570,16 +2595,43 @@ private: setenv("DTX", EnvOr("DTX", "0").c_str(), 1); setenv("OCTET_ALIGN", leg.octetAlign ? "1" : "0", 1); setenv("CODEC", leg.codec.c_str(), 1); + setenv("DTMF_PT", leg.dtmfPt ? std::to_string(*leg.dtmfPt).c_str() : "", 1); + setenv("DTMF_RX_PT", leg.dtmfRxPt ? std::to_string(*leg.dtmfRxPt).c_str() : "", 1); std::vector c; for (auto& s : argv) c.push_back(const_cast(s.c_str())); c.push_back(nullptr); execvp(c[0], c.data()); _exit(127); } + if (ctl[0] >= 0) close(ctl[0]); + if (pid < 0 && ctl[1] >= 0) { close(ctl[1]); ctl[1] = -1; } mediaPid_ = pid; + mediaCtl_ = ctl[1]; + } + + // The engine's SendDtmf action: one `dtmf ` line down the + // control pipe, whole or not at all — a write of at most PIPE_BUF bytes + // to a pipe is atomic, and the engine's DtmfMaxTones keeps the line + // under it (a longer line is refused here too, never half-written to + // be joined with the next). The leg may have died since the action was + // decided (its exit is reaped on this thread, after the actions ran): + // then the write fails with EPIPE — SIGPIPE is ignored in main() — and + // the tones are lost with the call, which the reap ends. + void SendDtmfToMedia(const std::string& tones) { + if (mediaCtl_ < 0) { Log(std::format("DTMF '{}' not sent: no control pipe to the media leg", tones)); return; } + const std::string line = std::format("dtmf {}\n", tones); + if (line.size() > PIPE_BUF) { Log(std::format("DTMF of {} tones not sent: longer than the control pipe's atomic write", tones.size())); return; } + const ssize_t n = write(mediaCtl_, line.data(), line.size()); + if (n != static_cast(line.size())) Log(std::format("DTMF '{}' not sent: {}", tones, n < 0 ? std::strerror(errno) : "short write to the media leg")); + } + + void CloseMediaCtl() { + if (mediaCtl_ >= 0) close(mediaCtl_); + mediaCtl_ = -1; } void StopMedia() { + CloseMediaCtl(); if (mediaPid_ <= 0) return; kill(mediaPid_, SIGTERM); for (int i = 0; i < 100; i++) { @@ -2599,6 +2651,7 @@ private: if (r == 0) return; // still running int code = WIFEXITED(status) ? WEXITSTATUS(status) : -1; mediaPid_ = -1; // reaped; don't re-terminate + CloseMediaCtl(); Execute(call_->OnMediaExit(code)); MaybeDropCall(); } @@ -2676,7 +2729,10 @@ void HandleMethodCall(GDBusConnection*, const gchar*, const gchar*, const gchar* return; } if (m == "SendDtmf") { - Log("SendDtmf accepted but not yet sent (TODO RFC4733)"); + const gchar* uni = nullptr; + const gchar* tones = nullptr; + g_variant_get(params, "(&s&s)", &uni, &tones); + TheEngine->SendDtmf(uni ? uni : "", tones ? tones : ""); g_dbus_method_invocation_return_value(inv, nullptr); return; } @@ -2740,6 +2796,11 @@ int main(int argc, char** argv) { std::println(std::cerr, "imsd: must run as root (xfrm / qmicli)"); return 1; } + // A write to the media leg's control pipe after the leg died must fail + // with EPIPE, not end the daemon (the client flow's sends carry + // MSG_NOSIGNAL; an inbound connection's reply is a plain write and is + // covered by this too). + signal(SIGPIPE, SIG_IGN); // The bring-up found nothing to do (IMSD_IDLE= in /run/imsd.env — the // script's arms: no SIM, a refused ims PDN): say so and diff --git a/implementations/media.cpp b/implementations/media.cpp index 7f37fa7..3d24966 100644 --- a/implementations/media.cpp +++ b/implementations/media.cpp @@ -26,11 +26,23 @@ DTX silence (every missing 20 ms slot is decoded as a NO_DATA frame for CNG/PLC; zeros for G.711). MIC=1 feeds live pw-record audio through the encoder. +DTMF goes out as RFC 4733 telephone-event packets at the payload type the +SDP negotiated for the codec's clock (DTMF_PT; unset = none): stdin is the +daemon's control pipe, and each `dtmf ` line queues its tones (0-9, +*, #, A-D). A tone takes seven 20 ms slots of the uplink in place of the +audio frames — the event's five (100 ms at -10 dBm0) and two in which its +end packet goes out again — then three slots of audio run before the next +tone. The far end's events — taken at DTMF_RX_PT (default DTMF_PT) and +at DTMF_PT, never at the audio's payload type — are played as the tones +they stand for, in step with the audio clock. + Usage: imsd-media imsd-media --selftest (payload pack/depay roundtrip, both formats, - both AMR codecs; G.711 table roundtrip) + both AMR codecs; G.711 table roundtrip; the + telephone-event payload, codes and Q.23 + frequency pairs) Env: CODEC MIC PLAY GAIN PLAY_GAIN AMR_MODE DTX MEDIA_TIMEOUT RTP_DUMP -OCTET_ALIGN AUDIO_USER MIC_SRC PCM_DUMP +OCTET_ALIGN AUDIO_USER MIC_SRC PCM_DUMP DTMF_PT DTMF_RX_PT (OCTET_ALIGN=0 selects RFC 4867 bandwidth-efficient payloads both ways; the daemon sets it from the negotiated SDP — 1 is the default/legacy behavior. AMR_MODE is the encoder mode of whichever AMR codec is active; default 2 @@ -45,6 +57,7 @@ with no PipeWire and no network.) #include #include #include +#include #include #include #include @@ -471,8 +484,13 @@ Child SpawnPw(bool play, bool toChild, int rate) { if (toChild) { dup2(pipefd[0], STDIN_FILENO); } else { dup2(pipefd[1], STDOUT_FILENO); } close(pipefd[0]); close(pipefd[1]); - int devnull = open("/dev/null", O_WRONLY); - if (devnull >= 0) { dup2(devnull, STDERR_FILENO); close(devnull); } + int devnull = open("/dev/null", O_RDWR); + if (devnull >= 0) { + dup2(devnull, STDERR_FILENO); + // our stdin is the daemon's control pipe: not pw-record's to hold + if (!toChild) dup2(devnull, STDIN_FILENO); + close(devnull); + } std::vector cargv; for (auto& s : argv) cargv.push_back(const_cast(s.c_str())); cargv.push_back(nullptr); @@ -507,19 +525,32 @@ struct TxState { sockaddr_storage latched{}; socklen_t latchedLen = 0; std::uint64_t tx = 0; std::uint64_t txMic = 0; + std::uint64_t txDtmf = 0; std::atomic micOn{false}; }; -void RtpSend(int sock, TxState& st, std::span payload) { +// One packet on the uplink, audio or event. Every packet stands in one +// 20 ms slot of the stream's single timestamp base (RFC 4733 §2.1: events +// share the audio's sequence and timestamp space), so the next slot's +// timestamp advances whichever kind went out; an audio frame is stamped +// with its slot (`ts` empty), an event packet with the event's start +// (`ts` given — constant over the event, §2.5.1.2). Returns the timestamp +// used. +std::uint32_t RtpSendPacket(int sock, TxState& st, int pt, bool marker, std::optional ts, std::span payload) { std::uint8_t hdr[12]; - sockaddr_storage a1, a2; socklen_t l1, l2; + sockaddr_storage a1; + sockaddr_storage a2; + socklen_t l1; + socklen_t l2; + std::uint32_t stamp; { std::scoped_lock g(st.lock); + stamp = ts.value_or(st.ts); hdr[0] = 0x80; - hdr[1] = static_cast(st.pt & 0x7F); + hdr[1] = static_cast((marker ? 0x80 : 0) | (pt & 0x7F)); hdr[2] = (st.seq >> 8) & 0xFF; hdr[3] = st.seq & 0xFF; - hdr[4] = (st.ts >> 24) & 0xFF; hdr[5] = (st.ts >> 16) & 0xFF; - hdr[6] = (st.ts >> 8) & 0xFF; hdr[7] = st.ts & 0xFF; + hdr[4] = (stamp >> 24) & 0xFF; hdr[5] = (stamp >> 16) & 0xFF; + hdr[6] = (stamp >> 8) & 0xFF; hdr[7] = stamp & 0xFF; hdr[8] = (st.ssrc >> 24) & 0xFF; hdr[9] = (st.ssrc >> 16) & 0xFF; hdr[10] = (st.ssrc >> 8) & 0xFF; hdr[11] = st.ssrc & 0xFF; st.seq = (st.seq + 1) & 0xFFFF; @@ -537,8 +568,126 @@ void RtpSend(int sock, TxState& st, std::span payload) { sendTo(a1, l1); // avoid double-send when latched == dst if (l2 && (l2 != l1 || std::memcmp(&a1, &a2, l1) != 0)) sendTo(a2, l2); + return stamp; } +// an audio frame: the stream's payload type, stamped with its slot +void RtpSend(int sock, TxState& st, std::span payload) { + RtpSendPacket(sock, st, st.pt, false, std::nullopt, payload); +} + +// ---- DTMF: RFC 4733 telephone-event packets on the uplink ----------------- +// A tone is one event of five slots (100 ms): an event packet in each in +// place of the audio frame, the duration growing by a slot, the marker bit +// on the first, the end bit on the fifth, which goes out three times, the +// repeats in two slots of their own (§2.5.1.4) — then three slots of audio +// before the next tone. The gateway's detector sees a 100 ms tone and a +// 100 ms gap: what a keypad press produces, and Q.24's 40 ms minimums +// with margin. The audio's own senders carry the events — per slot, a +// captured frame in the mic thread or a silence tick in the main loop +// without a mic (and the latch burst), `DtmfSender::SendSlot` sends +// either the event packet the slot carries or the frame, under one lock — +// so one packet leaves per slot, no frame can land inside an event +// whichever thread holds the slot, and the stream's clock stays the +// capture's; the slots an event took advanced it like frames +// (RtpSendPacket), so the audio that follows is stamped as if it had run +// through it. +constexpr int DtmfToneSlots = 5; +constexpr int DtmfEndRepeats = 2; +constexpr int DtmfGapSlots = 3; +constexpr std::uint8_t DtmfVolume = 10; // -10 dBm0 (§2.3.4: 0 to -63) + +// The event code of a keypad character (§3.2: 0-9, * = 10, # = 11, A-D = +// 12-15, either case); nullopt for anything else. +std::optional DtmfEvent(char c) { + if (c >= '0' && c <= '9') return c - '0'; + if (c == '*') return 10; + if (c == '#') return 11; + if (c >= 'A' && c <= 'D') return 12 + (c - 'A'); + if (c >= 'a' && c <= 'd') return 12 + (c - 'a'); + return std::nullopt; +} + +// §2.3: event, E | R | volume, the duration in timestamp units, big-endian. +std::array DtmfPayload(int event, bool end, std::uint16_t duration) { + return {static_cast(event), + static_cast((end ? 0x80 : 0) | DtmfVolume), + static_cast(duration >> 8), + static_cast(duration & 0xFF)}; +} + +// The tone an event stands for (ITU-T Q.23 / RFC 4733 §3.2): the keypad's +// row and column frequencies of events 0-15; nullopt for any other event. +struct DtmfPair { int low; int high; }; +std::optional DtmfFrequencies(int event) { + static constexpr std::array lows = {697, 770, 852, 941}; + static constexpr std::array highs = {1209, 1336, 1477, 1633}; + // keypad rows 1 2 3 A / 4 5 6 B / 7 8 9 C / * 0 # D, indexed by event + static constexpr std::array, 16> cell = {{ + {3, 1}, {0, 0}, {0, 1}, {0, 2}, {1, 0}, {1, 1}, {1, 2}, {2, 0}, + {2, 1}, {2, 2}, {3, 0}, {3, 2}, {0, 3}, {1, 3}, {2, 3}, {3, 3}}}; + if (event < 0 || event > 15) return std::nullopt; + const auto [row, col] = cell[static_cast(event)]; + return DtmfPair{lows[static_cast(row)], highs[static_cast(col)]}; +} + +// Sample `k` of a tone at `rate`: the two sines at `amp` each. +std::int16_t DtmfSample(DtmfPair f, std::uint32_t k, int rate, double amp) { + const double t = static_cast(k) / rate; + const double v = amp * (std::sin(2 * std::numbers::pi * f.low * t) + std::sin(2 * std::numbers::pi * f.high * t)); + return static_cast(std::lround(std::clamp(v, -32768.0, 32767.0))); +} + +// The tones the control pipe queued, paid out one slot at a time by the +// audio's own senders: one sender per slot, so no frame can land inside an +// event and the stream's clock is the capture's. +class DtmfSender { +public: + void Queue(std::string_view tones) { + std::scoped_lock g(lock_); + for (char c : tones) + if (auto ev = DtmfEvent(c)) queue_.push_back(*ev); + } + // Called by the audio's sender for every uplink slot, with the frame + // it would send: under one lock the slot is decided and sent — the + // event packet the slot carries (returns true: the frame stays unsent), + // or the frame (false; an empty frame, a DTX gap, sends nothing). One + // decision and one send per slot, so no frame can land inside an event + // whichever thread holds the slot. + bool SendSlot(int sock, TxState& st, int pt, std::span frame) { + std::scoped_lock g(lock_); + if (gap_ > 0) { + gap_--; + } else if (event_ >= 0 || !queue_.empty()) { + if (event_ < 0) { + event_ = queue_.front(); + queue_.pop_front(); + slot_ = 0; + start_.reset(); + } + slot_++; + const int reported = std::min(slot_, DtmfToneSlots); + const auto payload = DtmfPayload(event_, slot_ >= DtmfToneSlots, static_cast(reported * st.tsStep)); + start_ = RtpSendPacket(sock, st, pt, slot_ == 1, start_, payload); + { std::scoped_lock s(st.lock); st.txDtmf++; } + if (slot_ == DtmfToneSlots + DtmfEndRepeats) { + event_ = -1; + gap_ = DtmfGapSlots; + } + return true; + } + if (!frame.empty()) RtpSend(sock, st, frame); + return false; + } +private: + std::mutex lock_; + std::deque queue_; + int event_ = -1; // the event on the wire; -1 = none + int slot_ = 0; // its packets sent so far + int gap_ = 0; // slots of audio owed before the next event + std::optional start_; // the event's timestamp +}; + std::vector>> Depay(Codec c, std::span pl, bool octetAlign) { return octetAlign ? DepayOctet(c, pl) : DepayBe(c, pl); @@ -573,12 +722,14 @@ std::vector G711DecodeFrame(Codec c, std::span std::atomic Quit{false}; void OnSig(int) { Quit.store(true); } -// playout queue item -struct PktItem { std::uint32_t ts; std::vector payload; }; +// playout queue item: an audio packet, or (`event`) a telephone-event one +struct PktItem { std::uint32_t ts; std::vector payload; bool event = false; }; // --selftest: pack->depay roundtrip of every frame type, both payload // formats, both AMR codecs; G.711 digital zero, idempotence over the whole -// 16-bit range, and a 1 kHz sine surviving with the codec's nominal SNR. +// 16-bit range, and a 1 kHz sine surviving with the codec's nominal SNR; +// the RFC 4733 telephone-event payload bytes, every keypad character's +// event code, every event's Q.23 frequency pair and no tone past D. int SelfTest() { for (Codec c : {Codec::AmrWb, Codec::AmrNb}) { for (int ft = 0; ft < 16; ft++) { @@ -603,6 +754,42 @@ int SelfTest() { std::println(std::cerr, "selftest FAIL G.711 digital zero"); return 1; } + // RFC 4733 §2.3: event, E | R | volume (-10 dBm0), duration big-endian + if (DtmfPayload(1, false, 160) != std::array{0x01, 0x0A, 0x00, 0xA0} + || DtmfPayload(11, true, 1600) != std::array{0x0B, 0x8A, 0x06, 0x40}) { + std::println(std::cerr, "selftest FAIL telephone-event payload"); + return 1; + } + // §3.2: every keypad character's event code, the letters in either case, + // and no code for anything else + for (const char* keys : {"0123456789*#ABCD", "0123456789*#abcd"}) { + for (int ev = 0; ev < 16; ev++) { + if (DtmfEvent(keys[ev]) != ev) { + std::println(std::cerr, "selftest FAIL telephone-event code of '{}'", keys[ev]); + return 1; + } + } + } + if (DtmfEvent('E') || DtmfEvent('e') || DtmfEvent(' ') || DtmfEvent('\n')) { + std::println(std::cerr, "selftest FAIL telephone-event codes"); + return 1; + } + // Q.23: every event's pair — rows 697/770/852/941 Hz for 1 2 3 A, 4 5 6 B, + // 7 8 9 C, * 0 # D; columns 1209/1336/1477/1633 Hz — and no tone past D + static constexpr std::array, 16> q23 = {{ + {941, 1336}, {697, 1209}, {697, 1336}, {697, 1477}, {770, 1209}, {770, 1336}, {770, 1477}, {852, 1209}, + {852, 1336}, {852, 1477}, {941, 1209}, {941, 1477}, {697, 1633}, {770, 1633}, {852, 1633}, {941, 1633}}}; + for (int ev = 0; ev < 16; ev++) { + const auto f = DtmfFrequencies(ev); + if (!f || f->low != q23[static_cast(ev)].first || f->high != q23[static_cast(ev)].second) { + std::println(std::cerr, "selftest FAIL telephone-event tone {}", ev); + return 1; + } + } + if (DtmfFrequencies(16) || DtmfFrequencies(-1)) { + std::println(std::cerr, "selftest FAIL telephone-event tones past D"); + return 1; + } for (Codec c : {Codec::Pcma, Codec::Pcmu}) { for (int v = -32768; v <= 32767; v++) { std::uint8_t b = G711Encode(c, static_cast(v)); @@ -661,6 +848,8 @@ int main(int argc, char** argv) { bool rtpDump = EnvBool("RTP_DUMP", false); std::string micSrc = EnvOr("MIC_SRC", ""); bool pcmDump = EnvBool("PCM_DUMP", false); + const int dtmfPt = std::atoi(EnvOr("DTMF_PT", "0").c_str()); + const int dtmfRxPt = std::atoi(EnvOr("DTMF_RX_PT", EnvOr("DTMF_PT", "0").c_str()).c_str()); constexpr int ExitMediaTimeout = 3; constexpr int PrimeFrames = 8; constexpr int MaxFill = 25; @@ -693,8 +882,13 @@ int main(int argc, char** argv) { signal(SIGINT, OnSig); signal(SIGPIPE, SIG_IGN); + // ---- DTMF: the tones the control pipe (below) queues; the audio's + // senders pay them out slot by slot + DtmfSender dtmf; + // ---- mic uplink thread: pw-record (or the MIC_SRC file, real-time paced) - // through the codec's encoder, one RTP packet per 20 ms frame + // through the codec's encoder, one RTP packet per 20 ms frame — or the + // event packet whose slot it is std::atomic stop{false}; std::jthread micThread; if (mic) { @@ -731,23 +925,23 @@ int main(int argc, char** argv) { next += std::chrono::milliseconds(20); std::this_thread::sleep_until(next); } + st.micOn.store(true); std::memcpy(samples.data(), raw.data(), raw.size()); if (gain != 1.0) for (auto& sample : samples) { int v = static_cast(sample * gain); sample = static_cast(v < -32768 ? -32768 : (v > 32767 ? 32767 : v)); } - std::vector payload; + std::vector payload; // empty: a DTX gap, nothing to send in the slot if (amr) { std::vector frame = enc.Encode(samples.data(), amrMode, dtx); - if (frame.empty()) continue; - payload = PayloadFromFrame(codec, frame, octetAlign); + if (!frame.empty()) payload = PayloadFromFrame(codec, frame, octetAlign); } else { payload = G711Payload(codec, samples); } - st.micOn.store(true); + // the slot's packet: the event's, or this frame + if (dtmf.SendSlot(sock, st, dtmfPt, payload) || payload.empty()) continue; { std::scoped_lock g(st.lock); st.txMic++; } - RtpSend(sock, st, payload); } st.micOn.store(false); close(fd); @@ -783,6 +977,8 @@ int main(int argc, char** argv) { std::uint64_t cng = 0; std::uint64_t late = 0; std::uint64_t qdrop = 0; + std::uint64_t rxDtmf = 0; // telephone-event packets from the far end + std::uint64_t tones = 0; // events of theirs rendered as a tone std::jthread playThread; if (decodeOn) { playThread = std::jthread([&] { @@ -814,6 +1010,7 @@ int main(int argc, char** argv) { }; std::vector zero(frameSamples, 0); std::optional expect; + std::optional toneTs; // the event being rendered, by its start stamp for (;;) { PktItem item; { @@ -823,6 +1020,55 @@ int main(int argc, char** argv) { item = std::move(playq.front()); playq.pop_front(); } + if (item.event) { + // The far end's key press, played as the tone it stands + // for (§2.5.2.2 leaves rendering to the receiver). The + // gap before the event's start — a stock sender's DTX + // silence often precedes a key press (dev phone, + // 2026-10-15: 0 to 7 slots) — is filled as before an + // audio frame, so the tone starts where the event does. + // Each packet reports the event's duration so far from + // its start stamp; what the clock has not covered yet is + // synthesized — a retransmitted end packet adds nothing, + // a lost update is caught up by the next. A far end + // whose audio resumes past the event (the stock phone's + // does) meets the clock with the usual fill. + if (item.payload.size() < 4) continue; + const auto freqs = DtmfFrequencies(item.payload[0]); + if (!freqs) continue; + const std::uint32_t dur = (static_cast(item.payload[2]) << 8) | item.payload[3]; + const double amp = 16383.0 * std::pow(10.0, -(item.payload[1] & 0x3F) / 20.0); + if (!expect) { + for (int i = 0; i < PrimeFrames; i++) + if (!writePcm(zero)) return; + expect = item.ts; + } + const std::uint32_t before = (item.ts - *expect) & 0xFFFFFFFF; + if (before < 0x80000000u) { + int fill = static_cast(before / frameSamples); + if (fill > MaxFill) fill = 0; + for (int i = 0; i < fill; i++) { + cng++; + std::uint8_t nodata = 0x7C; + std::vector pcm = amr ? dec.Decode(&nodata, 1) : zero; + applyGain(pcm); + if (!writePcm(pcm)) return; + } + expect = item.ts; + } + const std::uint32_t end = (item.ts + dur) & 0xFFFFFFFF; + const std::uint32_t ahead = (end - *expect) & 0xFFFFFFFF; + if (ahead == 0 || ahead > static_cast(2 * rate)) continue; // covered, or not a tone's length + if (!toneTs || *toneTs != item.ts) { toneTs = item.ts; tones++; } + const std::uint32_t from = (*expect - item.ts) & 0xFFFFFFFF; + std::vector pcm(ahead); + for (std::uint32_t i = 0; i < ahead; i++) + pcm[i] = DtmfSample(*freqs, from + i, rate, amp); + applyGain(pcm); + if (!writePcm(pcm)) return; + expect = end; + continue; + } std::vector> frames; if (amr) { for (auto& [hdr, speech] : Depay(codec, item.payload, octetAlign)) { @@ -867,9 +1113,65 @@ int main(int argc, char** argv) { }); } + // ---- the control pipe that queues the tones: the daemon's + // `dtmf ` lines on stdin. No pipe (a hand run, /dev/null) ends + // the reader at once; the leg runs on. + std::jthread ctlThread([&] { + // a hand run in the background of a job-control shell: a read of + // its terminal would stop the whole process (SIGTTIN); blocked for + // this thread, the read fails with EIO and the reader ends + sigset_t ttin; + sigemptyset(&ttin); + sigaddset(&ttin, SIGTTIN); + pthread_sigmask(SIG_BLOCK, &ttin, nullptr); + std::string pending; + bool discarding = false; // an overlong line is dropped through its end + char buf[256]; + while (!stop.load()) { + pollfd pfd{STDIN_FILENO, POLLIN, 0}; + if (poll(&pfd, 1, 100) <= 0) continue; + const ssize_t n = read(STDIN_FILENO, buf, sizeof buf); + if (n <= 0) return; + std::string_view in(buf, static_cast(n)); + if (discarding) { + const std::size_t end = in.find('\n'); + if (end == std::string_view::npos) continue; + in.remove_prefix(end + 1); + discarding = false; + } + pending.append(in); + // a line of more than 4096 bytes with no end in sight: dropped + // through its end; the next line goes + if (pending.size() > 4096 && pending.find('\n') == std::string::npos) { + std::println(std::cerr, "imsd-media: control line of {} bytes without an end dropped", pending.size()); + pending.clear(); + discarding = true; + continue; + } + std::size_t nl; + while ((nl = pending.find('\n')) != std::string::npos) { + const std::string line = pending.substr(0, nl); + pending.erase(0, nl + 1); + // the same bound on a line whose end came in the read that + // took it past 4096 bytes + if (nl > 4096) { + std::println(std::cerr, "imsd-media: control line of {} bytes dropped", nl); + continue; + } + if (!line.starts_with("dtmf ")) { + std::println(std::cerr, "imsd-media: control line ignored: {}", line); + } else if (dtmfPt > 0) { + dtmf.Queue(std::string_view(line).substr(5)); + } else { + std::println(std::cerr, "imsd-media: {} dropped: no telephone-event payload type (DTMF_PT)", line); + } + } + } + }); + // ---- main recv loop std::vector silence = SilenceFrame(codec, 0, octetAlign); - for (int i = 0; i < 5; i++) RtpSend(sock, st, silence); // latch burst + for (int i = 0; i < 5; i++) dtmf.SendSlot(sock, st, dtmfPt, silence); // latch burst, the one path std::FILE* dump = rtpDump ? std::fopen((std::format("{}.rtp", out)).c_str(), "wb") : nullptr; double t0 = Now(); @@ -888,7 +1190,8 @@ int main(int argc, char** argv) { break; } if (now - lastTx >= 0.02 && !st.micOn.load()) { - RtpSend(sock, st, silence); + // no mic: the slot's packet is the event's, or a silence frame + dtmf.SendSlot(sock, st, dtmfPt, silence); lastTx = now; } std::uint8_t buf[65535]; @@ -926,7 +1229,14 @@ int main(int argc, char** argv) { std::fwrite(lb, 1, 4, dump); std::fwrite(buf, 1, static_cast(n), dump); } - if (playThread.joinable() && n >= 12 && (buf[1] & 0x7F) == pt) { + // the far end's events arrive at the payload type our own SDP gave + // telephone-event (DTMF_RX_PT), or at the one we send with when the + // answerer numbers for itself — never at the audio's: a frame is + // audio first + const int rxPt = n >= 12 ? buf[1] & 0x7F : -1; + const bool isEvent = n >= 12 && rxPt != pt && ((dtmfRxPt > 0 && rxPt == dtmfRxPt) || (dtmfPt > 0 && rxPt == dtmfPt)); + if (isEvent) rxDtmf++; + if (playThread.joinable() && n >= 12 && (rxPt == pt || isEvent)) { std::size_t off = RtpPayloadOffset(std::span(buf, static_cast(n))); std::uint32_t pktTs = (static_cast(buf[4]) << 24) | (buf[5] << 16) | (buf[6] << 8) | buf[7]; @@ -935,13 +1245,14 @@ int main(int argc, char** argv) { playq.pop_front(); qdrop++; } - playq.push_back({pktTs, std::vector(buf + off, buf + n)}); + playq.push_back({pktTs, std::vector(buf + off, buf + n), isEvent}); qcv.notify_one(); } } stop.store(true); qcv.notify_all(); + if (ctlThread.joinable()) ctlThread.join(); if (micThread.joinable()) micThread.join(); if (playThread.joinable()) playThread.join(); if (dump) std::fclose(dump); @@ -961,14 +1272,14 @@ int main(int argc, char** argv) { // .stats sidecar (tiny; always written) if (std::FILE* sf = std::fopen((std::format("{}.stats", out)).c_str(), "w")) { std::print(sf, - "{{\"codec\": \"{}\", \"tx\": {}, \"tx_mic\": {}, \"rx\": {}, \"rx_bytes\": {}, " - "\"rx_played\": {}, \"cng\": {}, \"late\": {}, \"qdrop\": {}, " - "\"first_src\": \"{}\", \"dst\": \"{}:{}\", \"pt\": {}, \"mic\": {}, " + "{{\"codec\": \"{}\", \"tx\": {}, \"tx_mic\": {}, \"tx_dtmf\": {}, \"rx\": {}, \"rx_bytes\": {}, " + "\"rx_played\": {}, \"rx_dtmf\": {}, \"tones\": {}, \"cng\": {}, \"late\": {}, \"qdrop\": {}, " + "\"first_src\": \"{}\", \"dst\": \"{}:{}\", \"pt\": {}, \"dtmf_pt\": {}, \"dtmf_rx_pt\": {}, \"mic\": {}, " "\"play\": {}, \"amr_mode\": {}, \"media_ended\": {}}}", - CodecName(codec), st.tx, st.txMic, rx, rxBytes, rxPlayed, cng, late, qdrop, firstSrc, - rIp, rPort, pt, mic, play, amrMode, mediaEnded); + CodecName(codec), st.tx, st.txMic, st.txDtmf, rx, rxBytes, rxPlayed, rxDtmf, tones, cng, late, qdrop, firstSrc, + rIp, rPort, pt, dtmfPt, dtmfRxPt, mic, play, amrMode, mediaEnded); std::fclose(sf); } - std::println("imsd-media: codec={} tx={} tx_mic={} rx={} rx_played={} cng={} late={} " "qdrop={} rx_bytes={} first_src={} media_ended={}", CodecName(codec), st.tx, st.txMic, rx, rxPlayed, cng, late, qdrop, rxBytes, firstSrc, mediaEnded); + std::println("imsd-media: codec={} tx={} tx_mic={} tx_dtmf={} rx={} rx_played={} rx_dtmf={} tones={} cng={} late={} " "qdrop={} rx_bytes={} first_src={} media_ended={}", CodecName(codec), st.tx, st.txMic, st.txDtmf, rx, rxPlayed, rxDtmf, tones, cng, late, qdrop, rxBytes, firstSrc, mediaEnded); return mediaEnded ? ExitMediaTimeout : 0; } diff --git a/interfaces/Imsd-Engine.cppm b/interfaces/Imsd-Engine.cppm index 52a4e1f..7c1d4ce 100644 --- a/interfaces/Imsd-Engine.cppm +++ b/interfaces/Imsd-Engine.cppm @@ -18,8 +18,10 @@ That keeps the whole call FSM — PRACK on reliable 18x, ACK of 2xx vs non-2xx finals, CANCEL/answer races, remote BYE, media-plane far-end-hangup, the emergency attempt chain (urn:service:sos first for a classified number, one fallback to a plain INVITE of the digits; a 380 Alternative Service upgrade -the other way), and the terminating (UAS) side: 100/180 on an inbound INVITE, -200-with-answer on Accept, 486 reject, the remote-CANCEL 200+487 pair — +the other way), the terminating (UAS) side: 100/180 on an inbound INVITE, +200-with-answer on Accept, 486 reject, the remote-CANCEL 200+487 pair — and +DTMF, which reaches the media leg only on an active call whose SDP +negotiated telephone-event, as the tones a keypad has — unit-testable against recorded network dialogs with no I/O (tests/Engine). State/reason strings are the frozen D-Bus vocabulary the dialer maps on @@ -280,12 +282,24 @@ export namespace imsd::engine { // Parsed SDP answer needed to bring up the media leg. `codec` is one of // imsd-media's: "AMR-WB", "AMR", "PCMA", "PCMU" (empty = none we play). + // `dtmfPt` is the telephone-event payload type at the codec's clock + // (RFC 4733) the leg sends with — the peer's SDP's; nullopt = none + // negotiated, no DTMF on this call. `dtmfRxPt` is a second number the + // leg takes events at: our own SDP's at that clock — our offer's + // number on a call we placed (an answerer sends with the offer's + // numbers, RFC 3264 §6.1, and may renumber ours in the answer), the + // offer's we echoed on a call we answered; the answer's telephone-event + // number instead when the answer gave the audio our offer's number, + // which is then the leg's audio number. The leg takes events at + // either number, never at the audio's. struct MediaLeg { std::string remoteIp; int remotePort = 0; int payloadType = -1; bool octetAlign = false; std::string codec = "AMR-WB"; + std::optional dtmfPt; + std::optional dtmfRxPt; bool Valid() const { return !remoteIp.empty() && remotePort > 0 && payloadType >= 0; } }; @@ -295,6 +309,7 @@ export namespace imsd::engine { SendResponse, // answer the just-handled inbound request on its origin channel StartMedia, // bring up the media leg described by `media` StopMedia, // tear the media leg down + SendDtmf, // hand `text` (the tones) to the running media leg State, // emit CallStateChanged(uni, `state`, `reason`) Deleted, // emit CallDeleted(uni); drop this machine SetDeadline, // arm a setup timer for `seconds`; fire calls OnDeadline() @@ -308,6 +323,11 @@ export namespace imsd::engine { double seconds = 0; }; + // The most tones one SendDtmf may carry (a keypad sends one; a dialer's + // string is a short sequence): the leg's `dtmf \n` line stays + // within one atomic pipe write, POSIX's 512-byte PIPE_BUF floor. + inline constexpr std::size_t DtmfMaxTones = 128; + // ring/answer budget, and how long to wait for a 487 after CANCEL — the // imsd.py INVITE_TIMEOUT / CANCEL_487_TIMEOUT constants. inline constexpr double InviteTimeout = 90.0; @@ -435,7 +455,7 @@ export namespace imsd::engine { spec.payloadType = leg.payloadType; spec.octetAlign = leg.octetAlign; spec.codec = leg.codec; - spec.dtmfPt = imsd::sdp::TelephoneEventPt(Body(invite_), imsd::sdp::CodecRate(leg.codec)); + spec.dtmfPt = leg.dtmfPt; localSdp_ = imsd::sdp::BuildAnswer(spec); // session timers: echo the caller's Session-Expires; with no // refresher stated, make the caller (uac) the refresher — its @@ -608,6 +628,45 @@ export namespace imsd::engine { return a; } + // The user pressed keys: the tones go to the media leg as RFC 4733 + // events on an active call whose SDP negotiated telephone-event at + // the codec's clock; otherwise they are dropped with a line that + // says why — there is no second method (no SIP INFO). + // Upper and lower case letters are one event; anything but 0-9, *, + // # and A-D drops the whole string rather than sending part of + // what was pressed, and so does a string past DtmfMaxTones — a + // keypad sends one key, a dialer's string a short sequence; the + // bound keeps the leg's control line within one atomic pipe write + // (PIPE_BUF), so a line is sent whole or not at all. + std::vector OnDtmf(std::string_view tones) { + std::vector a; + if (tones.size() > DtmfMaxTones) { + a.push_back(LogAct(std::format("call {}: DTMF of {} tones dropped: more than {}", uni_, tones.size(), DtmfMaxTones))); + return a; + } + std::string sent; + for (char c : tones) { + const char u = static_cast(std::toupper(static_cast(c))); + if (!((u >= '0' && u <= '9') || u == '*' || u == '#' || (u >= 'A' && u <= 'D'))) { + a.push_back(LogAct(std::format("call {}: DTMF '{}' dropped: '{}' is not a tone", uni_, tones, c))); + return a; + } + sent += u; + } + if (sent.empty()) return a; + if (state_ != CallState::Active) { + a.push_back(LogAct(std::format("call {}: DTMF '{}' dropped: the call is {}", uni_, sent, engine::StateStr(state_)))); + return a; + } + if (!dtmfPt_) { + a.push_back(LogAct(std::format("call {}: DTMF '{}' dropped: no telephone-event negotiated for this call", uni_, sent))); + return a; + } + a.push_back(LogAct(std::format("call {}: DTMF '{}' -> media leg (telephone-event pt {})", uni_, sent, *dtmfPt_))); + a.push_back(Action{Action::Type::SendDtmf, sent}); + return a; + } + // The shell is abandoning this machine (an emergency dial preempts // it while it waits on a CANCEL's 487): terminate bookkeeping only — // the CANCEL already sent stands, and the late final for this @@ -687,6 +746,7 @@ export namespace imsd::engine { bool userCancelled_ = false; // CANCEL came from OnHangup, not a deadline std::set pracked_; std::string answerSdp_; + std::optional dtmfPt_; // the running leg's telephone-event pt, set at media start // Fresh dialog identity for the current attempt. A fallback INVITE // after a final response is a new transaction AND a clean new dialog @@ -779,6 +839,13 @@ export namespace imsd::engine { leg.payloadType = ans.payloadType; leg.octetAlign = ans.octetAlign; leg.codec = ans.codec; + leg.dtmfPt = ans.dtmfPt; + // the second number the leg takes events at: our offer's on a call + // we placed — the answer's instead when the answer gave the audio + // that very number, which is then the leg's audio number + const int offered = imsd::sdp::OfferedTelephoneEventPt(imsd::sdp::CodecRate(leg.codec)); + leg.dtmfRxPt = !leg.dtmfPt ? std::nullopt + : std::optional(direction_ == Direction::Outgoing && offered != leg.payloadType ? offered : *leg.dtmfPt); return leg; } @@ -820,6 +887,7 @@ export namespace imsd::engine { return a; } Action StartMediaAct(const MediaLeg& leg) { + dtmfPt_ = leg.dtmfPt; Action a{Action::Type::StartMedia}; a.media = leg; return a; diff --git a/interfaces/Imsd-Sdp.cppm b/interfaces/Imsd-Sdp.cppm index 723720d..1c77dc1 100644 --- a/interfaces/Imsd-Sdp.cppm +++ b/interfaces/Imsd-Sdp.cppm @@ -9,7 +9,9 @@ The offer is by default an AMR-WB (octet-aligned) + AMR + telephone-event shape with an optional GSMA IR.92 QoS-precondition block; it is what commercial IMS cores expect from a VoLTE UE, and its exact bytes are pinned by tests/Sdp. The parser extracts only what the media plane needs: remote -address/port, the negotiated payload type, the codec and octet-align mode — it +address/port, the negotiated payload type, the codec and octet-align mode, and +the telephone-event payload type among the m-line's formats at that codec's +clock (DTMF, RFC 4733) — it reads answers to our offers and inbound offers alike (same fields either way). The codecs it knows are the ones imsd-media can play: AMR-WB, AMR (narrowband) and G.711 (PCMA/PCMU) — the last two are what a PSTN gateway offers when a @@ -82,6 +84,11 @@ export namespace imsd::sdp { std::vector codecs; // offered codecs in order; empty = AMR-WB, AMR }; + // The payload type our offer gives telephone-event at a clock rate: the + // number an answerer sends with (RFC 3264 §6.1), whatever the answer + // renumbered ours to. + int OfferedTelephoneEventPt(int clockRate) { return clockRate == 16000 ? 98 : 100; } + // Our offer: AMR-WB (octet-aligned) first, AMR narrowband second — the // 3GPP-mandatory codec every IMS core and PSTN gateway can answer, so a // call we place TO a landline no longer depends on the far side @@ -107,8 +114,8 @@ export namespace imsd::sdp { s += "t=0 0\r\n"; std::string pts; for (const auto& c : codecs) pts += std::format(" {}", pt(c)); - if (wb) pts += " 98"; - if (nb) pts += " 100"; + if (wb) pts += std::format(" {}", OfferedTelephoneEventPt(16000)); + if (nb) pts += std::format(" {}", OfferedTelephoneEventPt(8000)); s += std::format("m=audio {} RTP/AVP{}\r\n", o.rtpPort, pts); s += std::format("b=AS:{}\r\n", wb ? 41 : amr ? 30 : 80); s += "b=RS:512\r\n"; @@ -125,12 +132,12 @@ export namespace imsd::sdp { } } if (wb) { - s += "a=rtpmap:98 telephone-event/16000\r\n"; - s += "a=fmtp:98 0-15\r\n"; + s += std::format("a=rtpmap:{} telephone-event/16000\r\n", OfferedTelephoneEventPt(16000)); + s += std::format("a=fmtp:{} 0-15\r\n", OfferedTelephoneEventPt(16000)); } if (nb) { - s += "a=rtpmap:100 telephone-event/8000\r\n"; - s += "a=fmtp:100 0-15\r\n"; + s += std::format("a=rtpmap:{} telephone-event/8000\r\n", OfferedTelephoneEventPt(8000)); + s += std::format("a=fmtp:{} 0-15\r\n", OfferedTelephoneEventPt(8000)); } s += "a=ptime:20\r\n"; s += "a=maxptime:240\r\n"; @@ -150,22 +157,68 @@ export namespace imsd::sdp { int payloadType = -1; bool octetAlign = false; std::string codec = "AMR-WB"; // "AMR-WB" | "AMR" | "PCMA" | "PCMU" | "" (none we play) + std::optional dtmfPt; // telephone-event at the codec's clock rate; nullopt = no DTMF }; + // Sample rate of a codec we play: AMR-WB is 16 kHz, everything else + // (AMR, G.711) is narrowband 8 kHz. Also the RTP clock rate. + int CodecRate(std::string_view codec) { return codec == "AMR-WB" ? 16000 : 8000; } + constexpr std::string_view Avp = " RTP/AVP "; // after the m=audio port: the one profile we speak + + // The m=audio line's payload types in its order; empty without the line, + // its port or the " RTP/AVP " literal. + std::vector AudioFormats(std::string_view body) { + std::vector pts; + auto m = Find(body, "m=audio "); + if (!m) return pts; + std::string_view portTok = TokenAt(body, *m + 8); + std::size_t afterPort = *m + 8 + portTok.size(); + if (!ParseInt(portTok) || !body.substr(afterPort).starts_with(Avp)) return pts; + // digits + spaces up to CR/LF + std::size_t ptsBegin = afterPort + Avp.size(); + std::size_t ptsEnd = body.find_first_not_of("0123456789 ", ptsBegin); + std::string_view ptsRun = body.substr(ptsBegin, ptsEnd == std::string_view::npos ? std::string_view::npos : ptsEnd - ptsBegin); + for (auto part : std::views::split(ptsRun, ' ')) + if (auto v = ParseInt(std::string_view(part))) pts.push_back(*v); + return pts; + } + + // A payload type's rtpmap encoding, `NAME/CLOCK[/CHANNELS]` upper-cased + // (RFC 4566 §6: encoding names are case-insensitive); empty without one. + std::string Rtpmap(std::string_view body, int pt) { + const std::string key = std::format("a=rtpmap:{} ", pt); + auto at = Find(body, key); + if (!at) return ""; + std::size_t begin = *at + key.size(); + std::size_t end = body.find_first_of("\r\n", begin); + std::string enc(body.substr(begin, end == std::string_view::npos ? std::string_view::npos : end - begin)); + for (char& c : enc) + c = static_cast(std::toupper(static_cast(c))); + return enc; + } + + // Payload type of `telephone-event/` among the m=audio line's + // formats (DTMF, RFC 4733); nullopt when the line lists none. An rtpmap + // the line does not list is not negotiated: nothing is sent with a format + // the answer does not carry (RFC 3264 §8.3.2). The clock is matched + // whole: `/8000` is not `/80000`. + std::optional TelephoneEventPt(std::string_view body, int clockRate) { + const std::string needle = std::format("TELEPHONE-EVENT/{}", clockRate); + for (int pt : AudioFormats(body)) { + const std::string enc = Rtpmap(body, pt); + const bool whole = enc.size() == needle.size() || (enc.size() > needle.size() && (enc[needle.size()] == '/' || enc[needle.size()] == ' ')); + if (enc.starts_with(needle) && whole) return pt; + } + return std::nullopt; + } + // The encoding name an offer gives a payload type: its a=rtpmap line, // upper-cased (RFC 4566 §6: encoding names are case-insensitive), else // the static assignment for 0/8 (RFC 3551 §6 — a G.711 offer may carry // no rtpmap at all). Empty when neither applies. std::string CodecName(std::string_view body, int pt) { - std::string key = std::format("a=rtpmap:{} ", pt); - if (auto at = Find(body, key)) { - std::size_t nameBegin = *at + key.size(); - std::size_t nameEnd = body.find_first_of("/\r\n", nameBegin); - std::string name(body.substr(nameBegin, nameEnd == std::string_view::npos ? std::string_view::npos : nameEnd - nameBegin)); - for (char& c : name) - c = static_cast(std::toupper(static_cast(c))); - return name; - } + const std::string enc = Rtpmap(body, pt); + if (!enc.empty()) return enc.substr(0, enc.find('/')); if (pt == 0) return "PCMU"; if (pt == 8) return "PCMA"; return ""; @@ -202,7 +255,10 @@ export namespace imsd::sdp { // non-empty `prefs` replaces the codec order (and restricts acceptance // to the codecs named) — the CODECS override. With no acceptable codec // at all, the first pt is used and codec comes back EMPTY — pretending - // otherwise would defeat codec checks upstream. + // otherwise would defeat codec checks upstream. The telephone-event pt + // is the one at the chosen codec's clock (an event's rate is its rtpmap + // clock, RFC 4733 §2.4.1; an offer with two codec clocks lists + // telephone-event at each), so a tone goes out on the stream's own clock. Answer ParseAnswer(std::string_view body, std::span prefs = {}) { Answer a; if (auto at = Find(body, "c=IN IP6 ")) a.ip = TokenAt(body, *at + 9); @@ -214,17 +270,10 @@ export namespace imsd::sdp { auto port = ParseInt(portTok); // both the port and the " RTP/AVP " literal are required std::size_t afterPort = *m + 8 + portTok.size(); - constexpr std::string_view Avp = " RTP/AVP "; if (!port || !body.substr(afterPort).starts_with(Avp)) return a; a.port = *port; - // payload-type list: digits + spaces up to CR/LF - std::size_t ptsBegin = afterPort + Avp.size(); - std::size_t ptsEnd = body.find_first_not_of("0123456789 ", ptsBegin); - std::string_view ptsRun = body.substr(ptsBegin, ptsEnd == std::string_view::npos ? std::string_view::npos : ptsEnd - ptsBegin); - std::vector pts; - for (auto part : std::views::split(ptsRun, ' ')) - if (auto v = ParseInt(std::string_view(part))) pts.push_back(*v); + std::vector pts = AudioFormats(body); if (pts.empty()) return a; auto hasOctetAlign = [&](int p) { @@ -261,28 +310,10 @@ export namespace imsd::sdp { a.codec.clear(); } a.octetAlign = a.codec.starts_with("AMR") && hasOctetAlign(a.payloadType); + if (!a.codec.empty()) a.dtmfPt = TelephoneEventPt(body, CodecRate(a.codec)); return a; } - // Payload type of `telephone-event/` in a body's rtpmap lines - // (DTMF, RFC 4733); nullopt when the offer doesn't carry one. - std::optional TelephoneEventPt(std::string_view body, int clockRate) { - std::string needle = std::format("telephone-event/{}", clockRate); - std::size_t pos = 0; - while ((pos = body.find("a=rtpmap:", pos)) != std::string_view::npos) { - std::size_t ptBegin = pos + 9; - std::size_t lineEnd = body.find_first_of("\r\n", ptBegin); - std::size_t sp = body.find(' ', ptBegin); - if (sp != std::string_view::npos && (lineEnd == std::string_view::npos || sp < lineEnd)) { - auto pt = ParseInt(body.substr(ptBegin, sp - ptBegin)); - std::string_view name = body.substr(sp + 1, (lineEnd == std::string_view::npos ? body.size() : lineEnd) - sp - 1); - if (pt && name.starts_with(needle)) return pt; - } - pos = ptBegin; - } - return std::nullopt; - } - struct AnswerSpec { std::string local; // connection address (v4 or v6) int rtpPort = 0; @@ -293,14 +324,11 @@ export namespace imsd::sdp { std::optional dtmfPt; // the offer's telephone-event pt, if any }; - // Sample rate of a codec we play: AMR-WB is 16 kHz, everything else - // (AMR, G.711) is narrowband 8 kHz. Also the RTP clock rate. - int CodecRate(std::string_view codec) { return codec == "AMR-WB" ? 16000 : 8000; } - // SDP answer to an inbound audio offer: single codec at the offer's own // payload type, octet-align echoed only when the offer used it (absence // = bandwidth-efficient, RFC 4867 §8.1; G.711 has no fmtp at all), - // telephone-event kept when offered. b=AS is the codec's peak payload + // telephone-event kept when the offer's m-line lists one at the codec's + // clock (ParseAnswer's rule). b=AS is the codec's peak payload // rate plus IP/UDP/RTP overhead at 20 ms. Byte layout pinned by tests/Sdp. std::string BuildAnswer(const AnswerSpec& s) { std::string ipver = Is6(s.local) ? "IP6" : "IP4"; diff --git a/tests/Engine/main.cpp b/tests/Engine/main.cpp index 610dd79..75e1555 100644 --- a/tests/Engine/main.cpp +++ b/tests/Engine/main.cpp @@ -11,7 +11,10 @@ // reject -> 486, remote CANCEL -> 200+487, unusable offer -> 488 (naming // what was offered), a landline gateway's G.711-only and AMR-NB offers // ring and are answered at the offer's own payload types, ring timeout -> -// 480, BYE both ways. Then RegRecovery, the registration recovery policy: +// 480, BYE both ways. DTMF: the tones reach the media leg on an active +// call that negotiated telephone-event, and are dropped — with the reason +// — before the answer, after the call, without the negotiation or with a +// character that is no tone. Then RegRecovery, the registration recovery policy: // the refusal and silence thresholds, local refusals, the reconnect and // fresh backoffs, the throttle schedule, Retry-After, which waits a Dial // may cut short, when the old registration counts as gone, the holds a @@ -90,6 +93,21 @@ namespace { "v=0\r\no=- 1 1 IN IP6 2001:db8::9\r\ns=-\r\nc=IN IP6 2001:db8::9\r\n" "t=0 0\r\nm=audio 6004 RTP/AVP 97\r\na=rtpmap:97 AMR-WB/16000/1\r\n", callid); } + // A 200 OK around an arbitrary SDP answer, Content-Length from it — a + // message the shell's framing would deliver whole. + std::string R200Sdp(std::string_view callid, std::string_view sdp) { + return std::format( + "SIP/2.0 200 OK\r\n" + "Via: SIP/2.0/TCP [2001:db8::db43]:45061;branch=z9hG4bKx\r\n" + "From: ;tag=it\r\n" + "To: ;tag=remote7\r\n" + "Call-ID: {}\r\n" + "CSeq: 1 INVITE\r\n" + "Contact: \r\n" + "Content-Type: application/sdp\r\n" + "Content-Length: {}\r\n\r\n{}", callid, sdp.size(), sdp); + } + constexpr std::string_view SdpHead = "v=0\r\no=- 1 1 IN IP6 2001:db8::9\r\ns=-\r\nc=IN IP6 2001:db8::9\r\nt=0 0\r\n"; std::string Final(std::string_view callid, int code, std::string_view text) { return std::format( "SIP/2.0 {} {}\r\n" @@ -209,6 +227,7 @@ int main() { Check(media && media->media.remotePort == 6004, "media remote port"); Check(media && media->media.payloadType == 97, "media pt"); Check(media && media->media.codec == "AMR-WB", "media codec"); + Check(media && !media->media.dtmfPt, "an answer without telephone-event: the leg has no DTMF pt"); const Action* active = Find(a200, Action::Type::State); Check(active && active->state == "active" && active->reason == "accepted", "200 emits active/accepted"); Check(m.State() == CallState::Active, "state active"); @@ -220,6 +239,12 @@ int main() { Check(!Has(a200b, Action::Type::StartMedia) && !Has(a200b, Action::Type::State), "retransmitted 200: no second media start, no state action"); Check(m.State() == CallState::Active, "state still active after the retransmit"); + // DTMF on a call whose answer negotiated no telephone-event: dropped, said why + auto adt = m.OnDtmf("1"); + Check(!Has(adt, Action::Type::SendDtmf), "no telephone-event negotiated: no SendDtmf"); + const Action* dtLog = Find(adt, Action::Type::Log); + Check(dtLog && dtLog->text == "call ims-call-1: DTMF '1' dropped: no telephone-event negotiated for this call", "...and the line says so"); + // media leg exits code 3 (downlink dried up) -> BYE + terminated auto amx = m.OnMediaExit(3); Check(Sent(amx, "BYE ") != nullptr, "media far-end hangup sends BYE"); @@ -230,6 +255,73 @@ int main() { Check(m.Terminated(), "terminated"); } + // ---- DTMF, outgoing: the answer echoes our telephone-event/16000 at pt + // 98; the tones go to the leg once active, in upper case, and are + // dropped before the answer, with a character that is no tone, + // and after the call + { + imsd::util::Rng rng(43); + CallMachine m(c, rng, "ims-call-1d", "1233", 50004, false); + m.Start(); + auto early = m.OnDtmf("1"); + Check(!Has(early, Action::Type::SendDtmf), "DTMF while dialing: not sent"); + const Action* earlyLog = Find(early, Action::Type::Log); + Check(earlyLog && earlyLog->text == "call ims-call-1d: DTMF '1' dropped: the call is dialing", "...the line names the state"); + auto a200 = m.OnInviteResponse(R200Sdp(m.CallId(), std::format("{}m=audio 6004 RTP/AVP 97 98 100\r\na=rtpmap:97 AMR-WB/16000/1\r\na=rtpmap:98 telephone-event/16000\r\na=rtpmap:100 telephone-event/8000\r\n", SdpHead))); + const Action* media = Find(a200, Action::Type::StartMedia); + Check(media && media->media.dtmfPt == 98, "the leg carries the answer's telephone-event pt at the codec's clock (98, not the narrowband 100)"); + Check(media && media->media.dtmfRxPt == 98, "outgoing: the far end sends at our offer's number for that clock"); + Check(m.State() == CallState::Active, "active"); + auto sent = m.OnDtmf("12#*a"); + const Action* dt = Find(sent, Action::Type::SendDtmf); + Check(dt && dt->text == "12#*A", "active + negotiated: SendDtmf with the tones, letters upper-cased"); + const Action* dtLog = Find(sent, Action::Type::Log); + Check(dtLog && dtLog->text == "call ims-call-1d: DTMF '12#*A' -> media leg (telephone-event pt 98)", "...and the line names the pt"); + auto bad = m.OnDtmf("1x"); + Check(!Has(bad, Action::Type::SendDtmf), "a character that is no tone: nothing sent, not even the '1'"); + const Action* badLog = Find(bad, Action::Type::Log); + Check(badLog && badLog->text == "call ims-call-1d: DTMF '1x' dropped: 'x' is not a tone", "...the line names the character"); + Check(m.OnDtmf("").empty(), "an empty string: nothing, silently"); + const std::string atMost(DtmfMaxTones, '1'); + auto most = m.OnDtmf(atMost); + const Action* maxDt = Find(most, Action::Type::SendDtmf); + Check(maxDt && maxDt->text == atMost, "a string of DtmfMaxTones tones goes"); + auto tooMany = m.OnDtmf(std::format("{}1", atMost)); + Check(!Has(tooMany, Action::Type::SendDtmf) && Find(tooMany, Action::Type::Log) && Find(tooMany, Action::Type::Log)->text == std::format("call ims-call-1d: DTMF of {} tones dropped: more than {}", DtmfMaxTones + 1, DtmfMaxTones), "one more: dropped, the line names the bound"); + m.OnHangup(); + auto late = m.OnDtmf("5"); + Check(!Has(late, Action::Type::SendDtmf) && Find(late, Action::Type::Log) && Find(late, Action::Type::Log)->text.ends_with("the call is terminated"), "DTMF after the call: dropped, the line names the state"); + } + + // ---- DTMF, outgoing, the answer renumbered telephone-event (RFC 3264 + // allows it): we send with its number, the far end with our offer's + { + imsd::util::Rng rng(44); + CallMachine m(c, rng, "ims-call-1e", "1233", 50004, false); + m.Start(); + auto a200 = m.OnInviteResponse(R200Sdp(m.CallId(), std::format("{}m=audio 6004 RTP/AVP 97 101\r\na=rtpmap:97 AMR-WB/16000/1\r\na=rtpmap:101 telephone-event/16000\r\n", SdpHead))); + const Action* media = Find(a200, Action::Type::StartMedia); + Check(media && media->media.dtmfPt == 101 && media->media.dtmfRxPt == 98, "renumbered answer: send with 101, listen on our 98"); + // a narrowband answer: AMR at 99 with telephone-event/8000 at 102 + imsd::util::Rng rng2(45); + CallMachine n(c, rng2, "ims-call-1f", "1233", 50004, false); + n.Start(); + auto b200 = n.OnInviteResponse(R200Sdp(n.CallId(), std::format("{}m=audio 6004 RTP/AVP 99 102\r\na=rtpmap:99 AMR/8000/1\r\na=rtpmap:102 telephone-event/8000\r\n", SdpHead))); + const Action* nbMedia = Find(b200, Action::Type::StartMedia); + Check(nbMedia && nbMedia->media.codec == "AMR" && nbMedia->media.dtmfPt == 102 && nbMedia->media.dtmfRxPt == 100, "narrowband answer: send with its 102, listen on our offer's 100"); + // an answer numbering for itself that gives the audio our offer's + // telephone-event number: the receive number is the answer's, never + // the audio's (every downlink frame would be taken for an event) + imsd::util::Rng rng3(46); + CallMachine o(c, rng3, "ims-call-1g", "1233", 50004, false); + o.Start(); + auto c200 = o.OnInviteResponse(R200Sdp(o.CallId(), std::format("{}m=audio 6004 RTP/AVP 100 101\r\na=rtpmap:100 AMR/8000/1\r\na=rtpmap:101 telephone-event/8000\r\n", SdpHead))); + const Action* colMedia = Find(c200, Action::Type::StartMedia); + Check(colMedia && colMedia->media.payloadType == 100 && colMedia->media.dtmfPt == 101 && colMedia->media.dtmfRxPt == 101, "the audio at our telephone-event number: listen on the answer's 101, not the audio's 100"); + auto again = m.OnInviteResponse(R200(m.CallId())); + Check(!Find(again, Action::Type::StartMedia), "a retransmitted 200 starts no second leg"); + } + // ---- remote BYE mid-call { imsd::util::Rng rng(7); @@ -368,6 +460,12 @@ int main() { Check(ok && ok->text.contains("a=rtpmap:104 AMR-WB/16000/1\r\n"), "answer keeps AMR-WB at the offer's pt"); Check(ok && ok->text.contains("a=fmtp:104 octet-align=1\r\n"), "answer mirrors octet-align"); Check(ok && ok->text.contains("a=rtpmap:105 telephone-event/16000\r\n"), "answer keeps telephone-event"); + const Action* mtMedia = Find(aa, Action::Type::StartMedia); + Check(mtMedia && mtMedia->media.dtmfPt == 105, "incoming: the leg carries the offer's telephone-event pt"); + Check(mtMedia && mtMedia->media.dtmfRxPt == 105, "incoming: the far end sends at the number we echoed"); + auto mtDtmf = m.OnDtmf("9"); + const Action* mtDt = Find(mtDtmf, Action::Type::SendDtmf); + Check(mtDt && mtDt->text == "9", "incoming, accepted: the tones go to the leg"); Check(ok && ok->text.contains("Session-Expires: 1800;refresher=uac\r\n"), "200 echoes Session-Expires, caller refreshes"); Check(ok && ok->text.contains("Contact: nb = {"AMR"}; + Check(ParseAnswer(both, nb).dtmfPt == 106, "...and the narrowband one when AMR is chosen"); + Check(TelephoneEventPt(both, 8000) == 106, "listed on the m-line, the narrowband telephone-event pt is found"); + Check(!ParseAnswer(offer, nb).dtmfPt, "no codec we play chosen: no dtmfPt either"); + Check(!ParseAnswer("c=IN IP4 192.0.2.1\r\n" "m=audio 5000 RTP/AVP 8\r\n" "a=rtpmap:8 PCMA/8000\r\n").dtmfPt, "no telephone-event in the answer: dtmfPt empty"); + Check(!ParseAnswer("c=IN IP4 192.0.2.1\r\n" "m=audio 5000 RTP/AVP 8\r\n" "a=rtpmap:8 PCMA/8000\r\n" "a=rtpmap:101 telephone-event/8000\r\n").dtmfPt, "the answer dropped telephone-event from its m-line and kept the rtpmap: not negotiated (RFC 3264 §8.3.2: nothing is sent with a format the answer does not list)"); + Check(ParseAnswer("c=IN IP4 192.0.2.1\r\n" "m=audio 5000 RTP/AVP 8 101\r\n" "a=rtpmap:101 telephone-event/8000\r\n").dtmfPt == 101, "G.711 answer with telephone-event/8000: dtmfPt 101"); + Check(!ParseAnswer("c=IN IP4 192.0.2.1\r\n" "m=audio 5000 RTP/AVP 8 101\r\n" "a=rtpmap:101 telephone-event/16000\r\n").dtmfPt, "telephone-event at the wrong clock for the codec is not the call's"); + Check(ParseAnswer("c=IN IP4 192.0.2.1\r\n" "m=audio 5000 RTP/AVP 8 101\r\n" "a=rtpmap:101 TELEPHONE-EVENT/8000\r\n").dtmfPt == 101, "the name in another case (RFC 4566: case-insensitive)"); + Check(TelephoneEventPt("m=audio 1 RTP/AVP 98\r\na=rtpmap:98 Telephone-Event/16000/1\r\n", 16000) == 98, "mixed case, with a channel count"); + Check(!TelephoneEventPt("m=audio 1 RTP/AVP 98\r\na=rtpmap:98 telephone-event/80000\r\n", 8000), "the clock is matched whole: /80000 is not /8000"); Check(!TelephoneEventPt(offer, 48000).has_value(), "absent clock rate -> nullopt"); Check(!TelephoneEventPt("m=audio 1 RTP/AVP 0\r\n", 16000).has_value(), "no rtpmap -> nullopt"); }