mirror of
https://github.com/suitenumerique/meet.git
synced 2026-07-27 04:09:26 +00:00
✨(sip) add a sip software simulator to test unified encrypted flow
This commit is contained in:
Thomas Ramé
@@ -125,10 +125,16 @@ run-summary: ## start only the summary application and all needed services
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@$(COMPOSE) up --force-recreate -d celery-summary-summarize
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.PHONY: run-summary
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run-sip: ## start the SIP gateway and the browser-based softphone (Janus + nginx)
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@$(COMPOSE) up --force-recreate -d sip
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@$(COMPOSE) up --force-recreate -d sip-web-janus sip-web
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.PHONY: run-sip
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run:
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run: ## start the wsgi (production) and development server
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@$(MAKE) run-backend
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@$(MAKE) run-summary
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@$(MAKE) run-sip
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@$(COMPOSE) up --force-recreate -d frontend
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.PHONY: run
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+55
-2
@@ -168,8 +168,7 @@ services:
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- "3000:8080"
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dockerize:
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image: jwilder/dockerize
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platform: linux/x86_64
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image: powerman/dockerize:0.19.0
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crowdin:
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image: crowdin/cli:4.0.0
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@@ -246,6 +245,60 @@ services:
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depends_on:
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- redis
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# SIP gateway. Built from the sibling livekit-sip checkout so any dev
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# cloning both repos gets a working `docker compose up`. TODO: replace
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# the build: block with `image: livekit/sip` once upstream ships an
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# image with the team's video bridging.
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sip:
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build:
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context: ../livekit-sip
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dockerfile: build/sip/Dockerfile
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ports:
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- "5060:5060/udp"
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- "5060:5060/tcp"
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- "10000-10020:10000-10020/udp"
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environment:
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SIP_CONFIG_BODY: |
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api_key: 'devkey'
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api_secret: 'secret'
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ws_url: 'ws://livekit:7880'
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redis:
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address: 'redis:6379'
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sip_port: 5060
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rtp_port: 10000-10020
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use_external_ip: false
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logging:
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level: debug
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depends_on:
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- livekit
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- redis
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# Janus WebRTC gateway with the SIP plugin. Bridges a browser SIP demo
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# tab to the SIP gateway above. ICE-TCP enabled (janus.jcfg) so media
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# works under Lima port-forwarding without vmnet.
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sip-web-janus:
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build: ./docker/janus
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ports:
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- "10100-10120:10100-10120/tcp"
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volumes:
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- ./docker/janus/conf/janus.jcfg:/usr/local/etc/janus/janus.jcfg:ro
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- ./docker/janus/conf/janus.transport.http.jcfg:/usr/local/etc/janus/janus.transport.http.jcfg:ro
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- ./docker/janus/conf/janus.plugin.sip.jcfg:/usr/local/etc/janus/janus.plugin.sip.jcfg:ro
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depends_on:
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- sip
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# Serves docker/janus/web/{index.html,janus.js} and proxies /janus to
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# the Janus container so the demo is single-origin.
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sip-web:
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image: nginx:alpine
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ports:
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- "8088:80"
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volumes:
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- ./docker/janus/web:/usr/share/nginx/html:ro
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- ./docker/janus/conf/nginx.conf:/etc/nginx/conf.d/default.conf:ro
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depends_on:
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- sip-web-janus
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redis-summary:
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image: redis
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ports:
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@@ -0,0 +1,58 @@
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# Build Janus from source so ICE-TCP support is present (the canyan image
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# was compiled against libnice 0.1.16, which lacks the runtime ICE-TCP
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# capability Janus 1.1.x feature-detects at startup). Debian 13 ships
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# libnice 0.1.21+ which has ICE-TCP. Also: builds natively on arm64, so
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# no QEMU emulation overhead.
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FROM debian:13-slim AS builder
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ARG JANUS_VERSION=v1.1.4
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RUN apt-get update && apt-get install -y --no-install-recommends \
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build-essential ca-certificates git pkg-config \
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autoconf automake libtool gengetopt \
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libssl-dev libsrtp2-dev libglib2.0-dev libopus-dev libogg-dev \
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libcurl4-openssl-dev libconfig-dev libnice-dev \
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libmicrohttpd-dev libjansson-dev libsofia-sip-ua-dev \
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&& rm -rf /var/lib/apt/lists/*
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WORKDIR /src
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RUN git clone --depth 1 --branch ${JANUS_VERSION} \
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https://github.com/meetecho/janus-gateway.git janus
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WORKDIR /src/janus
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RUN ./autogen.sh && \
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./configure \
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--prefix=/opt/janus \
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--disable-rabbitmq --disable-mqtt --disable-nanomsg --disable-unix-sockets \
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--disable-data-channels \
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--disable-all-plugins \
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--enable-plugin-sip \
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--disable-all-handlers \
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--disable-all-transports --enable-rest && \
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make -j$(nproc) && \
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make install && \
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make configs
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FROM debian:13-slim
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# Debian 13 (trixie) renamed several libs in the time_t-64 transition:
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# libglib2.0-0t64, libcurl4t64. libconfig9 → libconfig11.
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RUN apt-get update && apt-get install -y --no-install-recommends \
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ca-certificates libssl3 libsrtp2-1 libglib2.0-0t64 libopus0 libogg0 \
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libcurl4t64 libconfig11 libnice10 libmicrohttpd12 libjansson4 \
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libsofia-sip-ua0 \
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&& rm -rf /var/lib/apt/lists/*
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COPY --from=builder /opt/janus /opt/janus
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# Make paths match canyan's so the rest of the compose mounts (configs, demos)
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# Just-Work without changes. The configs we mount in docker/janus/conf/ assume
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# /usr/local/etc/janus and /usr/local/lib/janus.
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RUN ln -s /opt/janus/bin/janus /usr/local/bin/janus && \
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ln -s /opt/janus/etc/janus /usr/local/etc/janus && \
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ln -s /opt/janus/lib/janus /usr/local/lib/janus && \
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ln -s /opt/janus/share/janus /usr/local/share/janus
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EXPOSE 8088 10100-10120/tcp 10100-10120/udp
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CMD ["/usr/local/bin/janus"]
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@@ -0,0 +1,42 @@
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// Minimal Janus config for browser-↔-SIP demo. Trimmed from the upstream
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// sample to the bits we actually need.
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general: {
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configs_folder = "/usr/local/etc/janus"
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plugins_folder = "/usr/local/lib/janus/plugins"
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transports_folder = "/usr/local/lib/janus/transports"
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events_folder = "/usr/local/lib/janus/events"
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log_to_stdout = true
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debug_level = 4
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server_name = "meet-sip-demo"
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}
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media: {
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// Janus ↔ Browser media (WebRTC over TCP — Lima only forwards TCP).
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rtp_port_range = "10100-10120"
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ipv6 = false
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}
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nat: {
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// The whole reason this works without vmnet — Janus offers TCP ICE
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// candidates the Mac browser can actually connect to.
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ice_tcp = true
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// libnice quirk: ICE-TCP needs ICE Lite mode on the server side or
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// the connectivity check state machine deadlocks. Janus warns about
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// this at startup if Lite is off while TCP is on.
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ice_lite = true
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// Janus advertises this IP in ICE candidates. From the Mac browser's
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// perspective, the Janus daemon lives at 127.0.0.1:<port> (Lima
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// forwards Mac:127.0.0.1:<port> → VM → docker port-map → container).
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nat_1_1_mapping = "127.0.0.1"
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keep_private_host = true
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}
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plugins: {
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// Only keep the SIP plugin. Everything else trimmed for boot time.
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disable = "libjanus_audiobridge.so,libjanus_videoroom.so,libjanus_streaming.so,libjanus_textroom.so,libjanus_recordplay.so,libjanus_voicemail.so,libjanus_echotest.so,libjanus_videocall.so,libjanus_nosip.so,libjanus_duktape.so,libjanus_lua.so"
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}
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transports: {
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// Only HTTP. WS is overkill for the demo and adds another port.
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disable = "libjanus_websockets.so,libjanus_mqtt.so,libjanus_nanomsg.so,libjanus_rabbitmq.so,libjanus_pfunix.so"
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}
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@@ -0,0 +1,8 @@
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// SIP plugin defaults. RTP range for the SIP leg (Janus ↔ livekit/sip)
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// is kept separate from WebRTC range — both legs are inside the docker
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// bridge network, so any UDP range works.
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general: {
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local_ip = "0.0.0.0"
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rtp_port_range = "20000-20100"
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events = true
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}
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@@ -0,0 +1,24 @@
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// HTTP transport: bind 0.0.0.0:8088 inside the container, served to the
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// browser through the nginx sidecar that also hosts the demo HTML.
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general: {
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json = "indented"
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base_path = "/janus"
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http = true
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port = 8088
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https = false
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mhd_connection_limit = 1020
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}
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admin: {
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admin_base_path = "/admin"
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admin_http = false
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admin_https = false
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}
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certificates: {
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}
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cors: {
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// Same-origin via nginx proxy; demo page and API share a host.
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# allow_origin = "*"
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}
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@@ -0,0 +1,33 @@
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server {
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listen 80;
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server_name _;
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# Docker's embedded DNS (127.0.0.11). Without this, nginx resolves
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# upstreams once at startup and breaks the moment a peer container is
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# recreated with a new bridge IP. The `valid=10s` plus a variable in
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# proxy_pass forces per-request re-resolution.
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resolver 127.0.0.11 valid=10s ipv6=off;
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# Serve the Janus demo HTML/JS extracted from the canyan image.
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root /usr/share/nginx/html;
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index index.html;
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# Janus HTTP API — same origin as the demo pages so the default
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# settings.js URL ("http://<host>:8088/janus") just works. Regex so
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# /janus and /janus/<session>/... proxy, but /janus.js still hits
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# the static file root.
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location ~ ^/janus(/|$) {
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# Variable upstream + the resolver above = re-DNS on each request.
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# Plain `proxy_pass http://sip-web-janus:8088` would cache the IP
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# for the lifetime of the worker process.
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set $janus_upstream sip-web-janus;
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proxy_pass http://$janus_upstream:8088;
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proxy_http_version 1.1;
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proxy_buffering off;
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proxy_read_timeout 120s;
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}
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location / {
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try_files $uri $uri/ =404;
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}
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}
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@@ -0,0 +1,405 @@
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<!DOCTYPE html>
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<!--
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Minimal SIP-via-Janus demo. Replaces the upstream siptest.html bundle with
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a single self-contained page. Everything is auto-driven: page load creates
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the Janus session, attaches to the SIP plugin, registers as guest, places
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the call. The user just needs to type the room PIN on the dialpad.
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Co-located only with janus.js (vendor lib, ~113KB) — no settings.js, no
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navbar/footer, no shared CSS, no logos. nginx serves these two files.
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-->
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<html lang="en">
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<head>
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<meta charset="utf-8">
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<title>SIP Demo</title>
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<style>
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* { box-sizing: border-box; }
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html, body { margin: 0; padding: 0; height: 100%; background: #1b1b1f; color: #e8e8e8;
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font: 14px/1.4 -apple-system, BlinkMacSystemFont, "Segoe UI", sans-serif; }
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#app { display: flex; flex-direction: column; height: 100vh; max-width: 720px; margin: 0 auto; padding: 16px; gap: 12px; }
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h1 { margin: 0; font-size: 18px; font-weight: 600; }
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#status { padding: 8px 12px; border-radius: 6px; background: #2c2c33; font-family: ui-monospace, monospace; font-size: 13px; }
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#video-wrap { position: relative; flex: 1; background: #000; border-radius: 8px; overflow: hidden; min-height: 240px; }
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#remoteVideo { width: 100%; height: 100%; object-fit: contain; }
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#noVideo { position: absolute; inset: 0; display: flex; align-items: center; justify-content: center; color: #555; font-size: 14px; }
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.row { display: flex; gap: 8px; flex-wrap: wrap; }
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button { font: inherit; padding: 8px 14px; border: 0; border-radius: 6px; background: #4a90e2; color: #fff; cursor: pointer; }
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button:hover:not(:disabled) { background: #357ec0; }
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button:disabled { opacity: 0.4; cursor: not-allowed; }
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button.danger { background: #dc3545; }
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button.danger:hover:not(:disabled) { background: #b62733; }
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button.success { background: #28a745; }
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button.success:hover:not(:disabled) { background: #1f8035; }
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.dialpad { display: grid; grid-template-columns: repeat(4, 1fr); gap: 6px; }
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.dialpad button { padding: 14px 0; font-size: 16px; background: #3a3a44; }
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.dialpad button:hover:not(:disabled) { background: #4a4a55; }
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#pin-row input { flex: 1; padding: 8px 12px; border: 1px solid #444; border-radius: 6px; background: #2c2c33; color: #fff; font: inherit; font-family: ui-monospace, monospace; letter-spacing: 0.2em; }
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#log { font-family: ui-monospace, monospace; font-size: 11px; max-height: 120px; overflow-y: auto; background: #15151a; padding: 8px; border-radius: 6px; color: #888; }
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.log-line { white-space: pre-wrap; word-break: break-word; }
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.log-err { color: #ff7373; }
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.disabled { opacity: 0.35; pointer-events: none; transition: opacity 0.2s; }
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</style>
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</head>
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<body>
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<div id="app">
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<h1>SIP demo (browser → Janus → livekit-sip → room)</h1>
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<div id="status">Loading…</div>
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<div id="video-wrap">
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<video id="remoteVideo" autoplay playsinline muted></video>
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<audio id="remoteAudio" autoplay></audio>
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<div id="noVideo">No remote video yet</div>
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</div>
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<div id="pin-row" class="row" style="display:none">
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<input type="text" id="pin" placeholder="Enter PIN then press # on the pad" inputmode="numeric" pattern="[0-9]*">
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<button id="sendPin" class="success">Send PIN</button>
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</div>
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<div id="dialpad" class="dialpad" style="display:none">
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<button data-d="1">1</button><button data-d="2">2</button><button data-d="3">3</button><button data-d="A">A</button>
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<button data-d="4">4</button><button data-d="5">5</button><button data-d="6">6</button><button data-d="B">B</button>
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<button data-d="7">7</button><button data-d="8">8</button><button data-d="9">9</button><button data-d="C">C</button>
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<button data-d="*">*</button><button data-d="0">0</button><button data-d="#">#</button><button data-d="D">D</button>
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</div>
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<div id="controls" class="row">
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<button id="muteBtn" class="danger">🔇 Mic muted</button>
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<button id="cameraBtn" class="success">📷 Camera on</button>
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<button id="speakerBtn" class="success">🔊 Speaker on</button>
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<button id="hangupBtn" class="danger" disabled>Hangup</button>
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<button id="rejoinBtn" style="display:none">Re-join</button>
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</div>
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||||
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<details><summary style="cursor:pointer; opacity:0.7; font-size:12px">debug log</summary>
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<div id="log"></div>
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</details>
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||||
</div>
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||||
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||||
<!-- webrtc-adapter is required by janus.js (cross-browser WebRTC API
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normalizer). The upstream siptest pulls it from CDN; we do the same.
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||||
Without it Janus.init() silently fails. -->
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||||
<script src="https://cdn.jsdelivr.net/npm/webrtc-adapter@9.0.1/out/adapter.min.js"></script>
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||||
<script src="janus.js"></script>
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||||
<script>
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||||
// ============================================================================
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||||
// Configuration (hard-coded; this is a dev demo)
|
||||
// ============================================================================
|
||||
const JANUS_SERVER = location.origin + "/janus"; // nginx proxies to janus
|
||||
const SIP_PROXY = "sip:sip:5060"; // docker service name
|
||||
const SIP_IDENTITY = "sip:demo@example.invalid"; // no registrar, guest
|
||||
const SIP_PEER = "sip:phone@sip:5060"; // user-part irrelevant
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// (Direct+PIN rule)
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||||
const DISPLAY_NAME = "test-user";
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const ICE_SERVERS = []; // ICE-TCP via janus.jcfg
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||||
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||||
// ============================================================================
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||||
// UI helpers
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||||
// ============================================================================
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||||
const $ = id => document.getElementById(id);
|
||||
function setStatus(text) { $("status").textContent = text; log(text); }
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||||
function log(msg, isErr = false) {
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||||
const line = document.createElement("div");
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||||
line.className = "log-line" + (isErr ? " log-err" : "");
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||||
line.textContent = "[" + new Date().toLocaleTimeString() + "] " + msg;
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||||
$("log").appendChild(line);
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||||
$("log").scrollTop = $("log").scrollHeight;
|
||||
}
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||||
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||||
// ============================================================================
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||||
// Janus state
|
||||
// ============================================================================
|
||||
let janus = null;
|
||||
let sipcall = null;
|
||||
let inCall = false;
|
||||
|
||||
// ============================================================================
|
||||
// Init: create Janus → session → SIP handle → register → call
|
||||
// ============================================================================
|
||||
Janus.init({
|
||||
debug: false,
|
||||
callback: () => {
|
||||
if (!Janus.isWebrtcSupported()) {
|
||||
setStatus("WebRTC not supported in this browser.");
|
||||
return;
|
||||
}
|
||||
createSession();
|
||||
}
|
||||
});
|
||||
|
||||
function createSession() {
|
||||
setStatus("Connecting to Janus…");
|
||||
janus = new Janus({
|
||||
server: JANUS_SERVER,
|
||||
iceServers: ICE_SERVERS,
|
||||
success: () => {
|
||||
setStatus("Connected. Attaching SIP plugin…");
|
||||
attachSip();
|
||||
},
|
||||
error: (err) => setStatus("Janus error: " + err),
|
||||
destroyed: () => setStatus("Janus session destroyed."),
|
||||
});
|
||||
}
|
||||
|
||||
function attachSip() {
|
||||
janus.attach({
|
||||
plugin: "janus.plugin.sip",
|
||||
opaqueId: "sip-demo-" + Janus.randomString(8),
|
||||
success: (handle) => {
|
||||
sipcall = handle;
|
||||
window.sipcall = handle; // for ad-hoc console debugging
|
||||
registerAsGuest();
|
||||
},
|
||||
error: (err) => setStatus("Plugin attach error: " + err),
|
||||
onmessage: handleMessage,
|
||||
onlocaltrack: (track, on) => {
|
||||
// We don't render local preview — Meet tab is already showing it.
|
||||
},
|
||||
onremotetrack: (track, mid, on, meta) => {
|
||||
if (!on) return;
|
||||
const stream = new MediaStream([track]);
|
||||
if (track.kind === "audio") {
|
||||
$("remoteAudio").srcObject = stream;
|
||||
} else if (track.kind === "video") {
|
||||
$("remoteVideo").srcObject = stream;
|
||||
$("noVideo").style.display = "none";
|
||||
}
|
||||
},
|
||||
oncleanup: () => { setStatus("Call cleaned up."); endCallUi(); },
|
||||
});
|
||||
}
|
||||
|
||||
function registerAsGuest() {
|
||||
setStatus("Registering as guest…");
|
||||
sipcall.send({
|
||||
message: {
|
||||
request: "register",
|
||||
type: "guest",
|
||||
proxy: SIP_PROXY,
|
||||
username: SIP_IDENTITY,
|
||||
display_name: DISPLAY_NAME,
|
||||
},
|
||||
});
|
||||
}
|
||||
|
||||
function placeCall() {
|
||||
setStatus("Calling " + SIP_PEER + "…");
|
||||
const tracks = [
|
||||
{ type: "audio", capture: true, recv: true },
|
||||
{ type: "video", capture: true, recv: true },
|
||||
];
|
||||
sipcall.createOffer({
|
||||
tracks: tracks,
|
||||
success: (jsep) => {
|
||||
sipcall.send({
|
||||
message: { request: "call", uri: SIP_PEER, autoaccept_reinvites: false },
|
||||
jsep: jsep,
|
||||
});
|
||||
},
|
||||
error: (err) => setStatus("createOffer failed: " + err),
|
||||
});
|
||||
}
|
||||
|
||||
// ============================================================================
|
||||
// Plugin event router
|
||||
// ============================================================================
|
||||
function handleMessage(msg, jsep) {
|
||||
const result = msg && msg.result;
|
||||
const event = result && result.event;
|
||||
if (event) log("event: " + event);
|
||||
|
||||
if (event === "registered") {
|
||||
setStatus("Registered. Placing call…");
|
||||
// Small delay to settle the registration
|
||||
setTimeout(placeCall, 100);
|
||||
}
|
||||
else if (event === "calling") {
|
||||
setStatus("Calling…");
|
||||
}
|
||||
else if (event === "accepted") {
|
||||
setStatus("Call accepted. You should hear the IVR — enter PIN on the dialpad.");
|
||||
startCallUi();
|
||||
if (jsep) sipcall.handleRemoteJsep({ jsep: jsep });
|
||||
}
|
||||
else if (event === "hangup") {
|
||||
setStatus("Call ended: " + (result.reason || result.code || ""));
|
||||
endCallUi();
|
||||
}
|
||||
else if (event === "missed_call") {
|
||||
setStatus("Missed call.");
|
||||
endCallUi();
|
||||
}
|
||||
else if (event === "registration_failed") {
|
||||
setStatus("Registration failed: " + result.code + " " + result.reason, true);
|
||||
}
|
||||
else if (msg && msg.error) {
|
||||
setStatus("Plugin error: " + msg.error, true);
|
||||
}
|
||||
|
||||
if (jsep && event !== "accepted") {
|
||||
sipcall.handleRemoteJsep({ jsep: jsep });
|
||||
}
|
||||
}
|
||||
|
||||
// ============================================================================
|
||||
// In-call UI: mute, hangup, dialpad, PIN entry
|
||||
// ============================================================================
|
||||
function startCallUi() {
|
||||
inCall = true;
|
||||
$("pin-row").style.display = "flex";
|
||||
$("pin-row").classList.remove("disabled");
|
||||
$("dialpad").style.display = "grid";
|
||||
$("dialpad").classList.remove("disabled");
|
||||
$("hangupBtn").disabled = false;
|
||||
$("rejoinBtn").style.display = "none";
|
||||
// Auto-mute mic so two tabs on the same Mac don't feedback-howl.
|
||||
if (sipcall.isAudioMuted && !sipcall.isAudioMuted()) {
|
||||
sipcall.muteAudio();
|
||||
}
|
||||
updateMuteButton();
|
||||
updateCameraButton();
|
||||
updateSpeakerButton();
|
||||
}
|
||||
|
||||
function endCallUi() {
|
||||
inCall = false;
|
||||
$("pin-row").style.display = "none";
|
||||
$("pin-row").classList.remove("disabled");
|
||||
$("dialpad").style.display = "none";
|
||||
$("dialpad").classList.remove("disabled");
|
||||
$("hangupBtn").disabled = true;
|
||||
$("rejoinBtn").style.display = "inline-block";
|
||||
$("remoteVideo").srcObject = null;
|
||||
$("remoteAudio").srcObject = null;
|
||||
$("noVideo").style.display = "flex";
|
||||
}
|
||||
|
||||
function updateMuteButton() {
|
||||
const b = $("muteBtn");
|
||||
if (!sipcall || !sipcall.isAudioMuted) return;
|
||||
if (sipcall.isAudioMuted()) {
|
||||
b.textContent = "🔇 Mic muted";
|
||||
b.className = "danger";
|
||||
} else {
|
||||
b.textContent = "🎤 Mic on";
|
||||
b.className = "success";
|
||||
}
|
||||
}
|
||||
|
||||
function updateCameraButton() {
|
||||
const b = $("cameraBtn");
|
||||
if (!sipcall || !sipcall.isVideoMuted) return;
|
||||
if (sipcall.isVideoMuted()) {
|
||||
b.textContent = "📷 Camera off";
|
||||
b.className = "danger";
|
||||
} else {
|
||||
b.textContent = "📷 Camera on";
|
||||
b.className = "success";
|
||||
}
|
||||
}
|
||||
|
||||
function updateSpeakerButton() {
|
||||
const b = $("speakerBtn");
|
||||
const audio = $("remoteAudio");
|
||||
if (audio.muted) {
|
||||
b.textContent = "🔇 Speaker off";
|
||||
b.className = "danger";
|
||||
} else {
|
||||
b.textContent = "🔊 Speaker on";
|
||||
b.className = "success";
|
||||
}
|
||||
}
|
||||
|
||||
$("muteBtn").addEventListener("click", () => {
|
||||
if (!sipcall) return;
|
||||
if (sipcall.isAudioMuted()) sipcall.unmuteAudio();
|
||||
else sipcall.muteAudio();
|
||||
setTimeout(updateMuteButton, 50);
|
||||
});
|
||||
|
||||
$("cameraBtn").addEventListener("click", () => {
|
||||
if (!sipcall || !sipcall.muteVideo) return;
|
||||
if (sipcall.isVideoMuted()) sipcall.unmuteVideo();
|
||||
else sipcall.muteVideo();
|
||||
setTimeout(updateCameraButton, 50);
|
||||
});
|
||||
|
||||
$("speakerBtn").addEventListener("click", () => {
|
||||
const audio = $("remoteAudio");
|
||||
audio.muted = !audio.muted;
|
||||
updateSpeakerButton();
|
||||
});
|
||||
|
||||
$("hangupBtn").addEventListener("click", () => {
|
||||
if (!sipcall) return;
|
||||
sipcall.send({ message: { request: "hangup" } });
|
||||
sipcall.hangup();
|
||||
});
|
||||
|
||||
$("rejoinBtn").addEventListener("click", () => location.reload());
|
||||
|
||||
// ============================================================================
|
||||
// DTMF — try the in-band RTP-event path first (RFC 4733 via the handle's
|
||||
// dtmf() method), which is what livekit-sip's IVR actually listens for.
|
||||
// Fall back to SIP INFO if RTP DTMF isn't available (older Janus, no
|
||||
// telephone-event in SDP, etc.).
|
||||
// ============================================================================
|
||||
function sendDtmf(digit) {
|
||||
if (!sipcall || !inCall) return;
|
||||
log("DTMF → " + digit);
|
||||
let sent = false;
|
||||
try {
|
||||
if (typeof sipcall.dtmf === "function") {
|
||||
sipcall.dtmf({
|
||||
dtmf: { tones: String(digit), duration: 200, gap: 70 },
|
||||
success: () => {},
|
||||
error: (err) => log("dtmf() error: " + err, true),
|
||||
});
|
||||
sent = true;
|
||||
}
|
||||
} catch (e) {
|
||||
log("dtmf() threw: " + e, true);
|
||||
}
|
||||
if (!sent) {
|
||||
// Fallback: SIP INFO method
|
||||
sipcall.send({
|
||||
message: { request: "dtmf_info", digit: String(digit) },
|
||||
});
|
||||
}
|
||||
}
|
||||
|
||||
document.querySelectorAll(".dialpad button").forEach(b => {
|
||||
b.addEventListener("click", () => sendDtmf(b.dataset.d));
|
||||
});
|
||||
|
||||
$("sendPin").addEventListener("click", () => {
|
||||
const pin = $("pin").value.replace(/[^\d#*]/g, "");
|
||||
if (!pin) return;
|
||||
const tones = (pin.endsWith("#") ? pin : pin + "#");
|
||||
log("DTMF batch → " + tones);
|
||||
// Send all tones in ONE call. Per-digit looping with setTimeout races
|
||||
// the WebRTC DTMF sender's internal queue and drops digits (we lost
|
||||
// the trailing "0" before "#" doing it digit-by-digit). The browser's
|
||||
// RTCDTMFSender paces them itself given duration/gap.
|
||||
try {
|
||||
sipcall.dtmf({
|
||||
dtmf: { tones: tones, duration: 400, gap: 100 },
|
||||
success: () => log("DTMF batch sent"),
|
||||
error: (err) => log("dtmf() error: " + err, true),
|
||||
});
|
||||
} catch (e) {
|
||||
log("dtmf() threw, falling back to SIP INFO per digit: " + e, true);
|
||||
for (const ch of tones) {
|
||||
sipcall.send({ message: { request: "dtmf_info", digit: String(ch) } });
|
||||
}
|
||||
}
|
||||
$("pin").value = "";
|
||||
// PIN + dialpad are no longer needed once you've submitted. Gateway
|
||||
// either bridges you in (correct PIN) or hangs up with wrong-pin
|
||||
// (the hangup event re-resets everything via endCallUi).
|
||||
$("pin-row").classList.add("disabled");
|
||||
$("dialpad").classList.add("disabled");
|
||||
});
|
||||
</script>
|
||||
</body>
|
||||
</html>
|
||||
File diff suppressed because it is too large
Load Diff
Reference in New Issue
Block a user