[Pkg-voip-commits] [janus] 14/282: First commit of SIPre plugin placeholder (WIP, still broken)
Jonas Smedegaard
dr at jones.dk
Wed Dec 20 21:53:23 UTC 2017
This is an automated email from the git hooks/post-receive script.
js pushed a commit to annotated tag debian/0.2.6-1
in repository janus.
commit a2bf959ef56ee5f804825e0276287d05d229cba7
Author: Lorenzo Miniero <lminiero at gmail.com>
Date: Fri Mar 17 17:57:38 2017 +0100
First commit of SIPre plugin placeholder (WIP, still broken)
---
Makefile.am | 10 +
conf/janus.plugin.sipre.cfg.sample | 22 +
configure.ac | 27 +-
html/demos.html | 22 +-
html/navbar.html | 3 +-
html/{siptest.html => sipretest.html} | 12 +-
html/sipretest.js | 603 +++++++
html/siptest.html | 10 +-
plugins/janus_sipre.c | 2937 +++++++++++++++++++++++++++++++++
utils.h | 11 +-
10 files changed, 3634 insertions(+), 23 deletions(-)
diff --git a/Makefile.am b/Makefile.am
index ce581e6..e3d43da 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -297,6 +297,16 @@ conf_DATA += conf/janus.plugin.sip.cfg.sample
EXTRA_DIST += conf/janus.plugin.sip.cfg.sample
endif
+if ENABLE_PLUGIN_SIPRE
+plugin_LTLIBRARIES += plugins/libjanus_sipre.la
+plugins_libjanus_sipre_la_SOURCES = plugins/janus_sipre.c
+plugins_libjanus_sipre_la_CFLAGS = $(plugins_cflags) $(LIBRE_CFLAGS)
+plugins_libjanus_sipre_la_LDFLAGS = $(plugins_ldflags) $(LIBRE_LDFLAGS) $(LIBRE_LIBS)
+plugins_libjanus_sipre_la_LIBADD = $(plugins_libadd) $(LIBRE_LIBADD)
+conf_DATA += conf/janus.plugin.sipre.cfg.sample
+EXTRA_DIST += conf/janus.plugin.sipre.cfg.sample
+endif
+
if ENABLE_PLUGIN_STREAMING
plugin_LTLIBRARIES += plugins/libjanus_streaming.la
plugins_libjanus_streaming_la_SOURCES = plugins/janus_streaming.c
diff --git a/conf/janus.plugin.sipre.cfg.sample b/conf/janus.plugin.sipre.cfg.sample
new file mode 100644
index 0000000..f7c76a6
--- /dev/null
+++ b/conf/janus.plugin.sipre.cfg.sample
@@ -0,0 +1,22 @@
+[general]
+; Specify which local IP address to use. If not set it will be automatically
+; guessed from the system
+;local_ip = 1.2.3.4
+
+; Enable local keep-alives to keep the registration open. Keep-alives are
+; sent in the form of OPTIONS requests, at the given interval inseconds.
+; (0 to disable)
+keepalive_interval = 120
+
+; Indicate if the server is behind NAT. If so, the server will use STUN
+; to guess its own public IP address and use it in the Contact header of
+; outgoing requests
+behind_nat = no
+
+; User-Agent string to be used
+; user_agent = Cool WebRTC Gateway
+; Expiration time for registrations
+register_ttl = 3600
+
+; Whether events should be sent to event handlers (default is yes)
+;events = no
diff --git a/configure.ac b/configure.ac
index b4fcb91..256d851 100644
--- a/configure.ac
+++ b/configure.ac
@@ -408,6 +408,12 @@ AC_ARG_ENABLE([plugin-sip],
[],
[enable_plugin_sip=maybe])
+AC_ARG_ENABLE([plugin-sipre],
+ [AS_HELP_STRING([--disable-plugin-sipre],
+ [Disable sipre plugin])],
+ [],
+ [enable_plugin_sipre=maybe])
+
AC_ARG_ENABLE([plugin-streaming],
[AS_HELP_STRING([--disable-plugin-streaming],
[Disable streaming plugin])],
@@ -451,6 +457,19 @@ PKG_CHECK_MODULES([SOFIA],
AC_SUBST([SOFIA_CFLAGS])
AC_SUBST([SOFIA_LIBS])
+PKG_CHECK_MODULES([LIBRE],
+ [libre],
+ [
+ AS_IF([test "x$enable_plugin_sipre" = "xmaybe"],
+ [enable_plugin_sipre=yes])
+ ],
+ [
+ AS_IF([test "x$enable_plugin_sipre" = "xyes"],
+ [AC_MSG_ERROR([libre not found. See README.md for installation instructions or use --disable-plugin-sip])])
+ ])
+AC_SUBST([SOFIA_CFLAGS])
+AC_SUBST([SOFIA_LIBS])
+
PKG_CHECK_MODULES([OPUS],
[opus],
[
@@ -481,6 +500,7 @@ AM_CONDITIONAL([ENABLE_PLUGIN_AUDIOBRIDGE], [test "x$enable_plugin_audiobridge"
AM_CONDITIONAL([ENABLE_PLUGIN_ECHOTEST], [test "x$enable_plugin_echotest" = "xyes"])
AM_CONDITIONAL([ENABLE_PLUGIN_RECORDPLAY], [test "x$enable_plugin_recordplay" = "xyes"])
AM_CONDITIONAL([ENABLE_PLUGIN_SIP], [test "x$enable_plugin_sip" = "xyes"])
+AM_CONDITIONAL([ENABLE_PLUGIN_SIPRE], [test "x$enable_plugin_sipre" = "xyes"])
AM_CONDITIONAL([ENABLE_PLUGIN_STREAMING], [test "x$enable_plugin_streaming" = "xyes"])
AM_CONDITIONAL([ENABLE_PLUGIN_VIDEOCALL], [test "x$enable_plugin_videocall" = "xyes"])
AM_CONDITIONAL([ENABLE_PLUGIN_VIDEOROOM], [test "x$enable_plugin_videoroom" = "xyes"])
@@ -587,8 +607,11 @@ AM_COND_IF([ENABLE_PLUGIN_VIDEOCALL],
[echo " Video Call: yes"],
[echo " Video Call: no"])
AM_COND_IF([ENABLE_PLUGIN_SIP],
- [echo " SIP Gateway: yes"],
- [echo " SIP Gateway: no"])
+ [echo " SIP Gateway (Sofia): yes"],
+ [echo " SIP Gateway (Sofia): no"])
+AM_COND_IF([ENABLE_PLUGIN_SIPRE],
+ [echo " SIP Gateway (libre): yes"],
+ [echo " SIP Gateway (libre): no"])
AM_COND_IF([ENABLE_PLUGIN_AUDIOBRIDGE],
[echo " Audio Bridge: yes"],
[echo " Audio Bridge: no"])
diff --git a/html/demos.html b/html/demos.html
index 1b9214e..b5593c6 100644
--- a/html/demos.html
+++ b/html/demos.html
@@ -32,11 +32,11 @@
<div class="page-header">
<h1>Janus WebRTC Gateway: Demo Tests</h1>
</div>
- <table class="table">
+ <table class="table table-striped">
<tr>
<td colspan=2><h3>Plugin demos</h3></td>
</tr>
- <tr class="active">
+ <tr>
<td><a href="echotest.html">Echo Test</a></td>
<td>A simple Echo Test demo, with knobs to control the bitrate.</td>
</tr>
@@ -44,15 +44,19 @@
<td><a href="streamingtest.html">Streaming</a></td>
<td>A media Streaming demo, with sample live and on-demand streams.</td>
</tr>
- <tr class="active">
+ <tr>
<td><a href="videocalltest.html">Video Call</a></td>
<td>A Video Call demo, a bit like AppRTC but with media passing through the gateway.</td>
</tr>
<tr>
- <td><a href="siptest.html">SIP Gateway</a></td>
+ <td><a href="siptest.html">SIP Gateway (Sofia)</a></td>
<td>A SIP Gateway demo, allowing you to register at a SIP server and start/receive calls.</td>
</tr>
- <tr class="active">
+ <tr>
+ <td><a href="sipretest.html">SIP Gateway (libre)</a></td>
+ <td>Same as the above, but using the libre-based plugin instead of the Sofia-based one.</td>
+ </tr>
+ <tr>
<td><a href="videoroomtest.html">Video Room</a></td>
<td>A videoconferencing demo, allowing you to join a video room with up to six users.</td>
</tr>
@@ -60,7 +64,7 @@
<td><a href="audiobridgetest.html">Audio Room</a></td>
<td>An audio mixing/bridge demo, allowing you join an Audio Room room.</td>
</tr>
- <tr class="active">
+ <tr>
<td><a href="textroomtest.html">Text Room</a></td>
<td>A text room demo, using DataChannels only.</td>
</tr>
@@ -68,7 +72,7 @@
<td><a href="voicemailtest.html">Voice Mail</a></td>
<td>A simple audio recorder demo, returning an .opus file after 10 seconds.</td>
</tr>
- <tr class="active">
+ <tr>
<td><a href="recordplaytest.html">Recorder/Playout</a></td>
<td>A demo to record audio/video messages, and subsequently replay them through WebRTC.</td>
</tr>
@@ -77,11 +81,11 @@
<td>A webinar-like screen sharing session, based on the Video Room plugin.</td>
</tr>
</table>
- <table class="table">
+ <table class="table table-striped">
<tr>
<td colspan=2><h3>Other demos</h3></td>
</tr>
- <tr class="active">
+ <tr>
<td><a href="devicetest.html">Device Selection</a></td>
<td>A variant of the Echo Test demo, that allows you to choose a specific capture device.</td>
</tr>
diff --git a/html/navbar.html b/html/navbar.html
index ee8a005..9769bde 100644
--- a/html/navbar.html
+++ b/html/navbar.html
@@ -17,7 +17,8 @@
<li><a href="echotest.html">Echo Test</a></li>
<li><a href="streamingtest.html">Streaming</a></li>
<li><a href="videocalltest.html">Video Call</a></li>
- <li><a href="siptest.html">SIP Gateway</a></li>
+ <li><a href="siptest.html">SIP Gateway (Sofia)</a></li>
+ <li><a href="sipretest.html">SIP Gateway (libre)</a></li>
<li><a href="videoroomtest.html">Video Room</a></li>
<li><a href="audiobridgetest.html">Audio Room</a></li>
<li><a href="textroomtest.html">Text Room</a></li>
diff --git a/html/siptest.html b/html/sipretest.html
similarity index 93%
copy from html/siptest.html
copy to html/sipretest.html
index d1dcfd8..e0c33a6 100644
--- a/html/siptest.html
+++ b/html/sipretest.html
@@ -4,7 +4,7 @@
<meta charset="utf-8">
<meta name="viewport" content="width=device-width, initial-scale=1.0"/>
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8" />
-<title>Janus WebRTC Gateway: SIP Gateway Demo</title>
+<title>Janus WebRTC Gateway: SIP Gateway Demo (libre)</title>
<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/webrtc-adapter/3.1.5/adapter.min.js" ></script>
<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/jquery/1.7.2/jquery.min.js" ></script>
<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/jquery.blockUI/2.70/jquery.blockUI.min.js" ></script>
@@ -13,7 +13,7 @@
<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/spin.js/2.3.2/spin.min.js"></script>
<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/blueimp-md5/2.6.0/js/md5.min.js"></script>
<script type="text/javascript" src="janus.js" ></script>
-<script type="text/javascript" src="siptest.js"></script>
+<script type="text/javascript" src="sipretest.js"></script>
<script>
$(function() {
$(".navbar-static-top").load("navbar.html", function() {
@@ -38,7 +38,7 @@
<div class="row">
<div class="col-md-12">
<div class="page-header">
- <h1>Plugin Demo: SIP Gateway
+ <h1>Plugin Demo: SIP Gateway (libre)
<button class="btn btn-default" autocomplete="off" id="start">Start</button>
</h1>
</div>
@@ -48,7 +48,11 @@
<h3>Demo details</h3>
<p>This demo shows how you can make use of the SIP plugin to interact with a
SIP Proxy (e.g., Kamailio) or PBX (e.g., Asterisk) in order to place or
- receive calls to and from other SIP clients.</p>
+ receive calls to and from other SIP clients. Specifically, it uses the libre-based
+ SIP plugin: in case you're interested in the Sofia-based one, check
+ <a href="siptest.html">this other demo</a> instead. Notice that both
+ plugins only exchange SIP messages from within the plugin itself: no SIP
+ is done in JavaScript, except for references to SIP URIs.</p>
<p>When started, the demo will allow you to insert a minimum set of information
required to REGISTER the web page as a SIP client at a SIP Proxy or PBX you specify.
This will allow you to call SIP URIs, or receive calls through the SIP Server itself.
diff --git a/html/sipretest.js b/html/sipretest.js
new file mode 100644
index 0000000..6ef163d
--- /dev/null
+++ b/html/sipretest.js
@@ -0,0 +1,603 @@
+// We make use of this 'server' variable to provide the address of the
+// REST Janus API. By default, in this example we assume that Janus is
+// co-located with the web server hosting the HTML pages but listening
+// on a different port (8088, the default for HTTP in Janus), which is
+// why we make use of the 'window.location.hostname' base address. Since
+// Janus can also do HTTPS, and considering we don't really want to make
+// use of HTTP for Janus if your demos are served on HTTPS, we also rely
+// on the 'window.location.protocol' prefix to build the variable, in
+// particular to also change the port used to contact Janus (8088 for
+// HTTP and 8089 for HTTPS, if enabled).
+// In case you place Janus behind an Apache frontend (as we did on the
+// online demos at http://janus.conf.meetecho.com) you can just use a
+// relative path for the variable, e.g.:
+//
+// var server = "/janus";
+//
+// which will take care of this on its own.
+//
+//
+// If you want to use the WebSockets frontend to Janus, instead, you'll
+// have to pass a different kind of address, e.g.:
+//
+// var server = "ws://" + window.location.hostname + ":8188";
+//
+// Of course this assumes that support for WebSockets has been built in
+// when compiling the gateway. WebSockets support has not been tested
+// as much as the REST API, so handle with care!
+//
+//
+// If you have multiple options available, and want to let the library
+// autodetect the best way to contact your gateway (or pool of gateways),
+// you can also pass an array of servers, e.g., to provide alternative
+// means of access (e.g., try WebSockets first and, if that fails, fall
+// back to plain HTTP) or just have failover servers:
+//
+// var server = [
+// "ws://" + window.location.hostname + ":8188",
+// "/janus"
+// ];
+//
+// This will tell the library to try connecting to each of the servers
+// in the presented order. The first working server will be used for
+// the whole session.
+//
+var server = null;
+if(window.location.protocol === 'http:')
+ server = "http://" + window.location.hostname + ":8088/janus";
+else
+ server = "https://" + window.location.hostname + ":8089/janus";
+
+var janus = null;
+var sipcall = null;
+var opaqueId = "sipretest-"+Janus.randomString(12);
+
+var started = false;
+var spinner = null;
+
+var selectedApproach = null;
+var registered = false;
+
+var incoming = null;
+
+
+$(document).ready(function() {
+ // Initialize the library (all console debuggers enabled)
+ Janus.init({debug: "all", callback: function() {
+ // Use a button to start the demo
+ $('#start').click(function() {
+ if(started)
+ return;
+ started = true;
+ $(this).attr('disabled', true).unbind('click');
+ // Make sure the browser supports WebRTC
+ if(!Janus.isWebrtcSupported()) {
+ bootbox.alert("No WebRTC support... ");
+ return;
+ }
+ // Create session
+ janus = new Janus(
+ {
+ server: server,
+ success: function() {
+ // Attach to echo test plugin
+ janus.attach(
+ {
+ plugin: "janus.plugin.sipre",
+ opaqueId: opaqueId,
+ success: function(pluginHandle) {
+ $('#details').remove();
+ sipcall = pluginHandle;
+ Janus.log("Plugin attached! (" + sipcall.getPlugin() + ", id=" + sipcall.getId() + ")");
+ // Prepare the username registration
+ $('#sipcall').removeClass('hide').show();
+ $('#login').removeClass('hide').show();
+ $('#registerlist a').unbind('click').click(function() {
+ selectedApproach = $(this).attr("id");
+ $('#registerset').html($(this).html()).parent().removeClass('open');
+ if(selectedApproach === "guest") {
+ $('#password').empty().attr('disabled', true);
+ } else {
+ $('#password').removeAttr('disabled');
+ }
+ switch(selectedApproach) {
+ case "secret":
+ bootbox.alert("Using this approach you'll provide a plain secret to REGISTER");
+ break;
+ case "ha1secret":
+ bootbox.alert("Using this approach might not work with Asterisk because the generated HA1 secret could have the wrong realm");
+ break;
+ case "guest":
+ bootbox.alert("Using this approach you'll try to REGISTER as a guest, that is without providing any secret");
+ break;
+ default:
+ break;
+ }
+ return false;
+ });
+ $('#register').click(registerUsername);
+ $('#server').focus();
+ $('#start').removeAttr('disabled').html("Stop")
+ .click(function() {
+ $(this).attr('disabled', true);
+ janus.destroy();
+ });
+ },
+ error: function(error) {
+ Janus.error(" -- Error attaching plugin...", error);
+ bootbox.alert(" -- Error attaching plugin... " + error);
+ },
+ consentDialog: function(on) {
+ Janus.debug("Consent dialog should be " + (on ? "on" : "off") + " now");
+ if(on) {
+ // Darken screen and show hint
+ $.blockUI({
+ message: '<div><img src="up_arrow.png"/></div>',
+ css: {
+ border: 'none',
+ padding: '15px',
+ backgroundColor: 'transparent',
+ color: '#aaa',
+ top: '10px',
+ left: (navigator.mozGetUserMedia ? '-100px' : '300px')
+ } });
+ } else {
+ // Restore screen
+ $.unblockUI();
+ }
+ },
+ onmessage: function(msg, jsep) {
+ Janus.debug(" ::: Got a message :::");
+ Janus.debug(JSON.stringify(msg));
+ // Any error?
+ var error = msg["error"];
+ if(error != null && error != undefined) {
+ if(!registered) {
+ $('#server').removeAttr('disabled');
+ $('#username').removeAttr('disabled');
+ $('#displayname').removeAttr('disabled');
+ $('#password').removeAttr('disabled');
+ $('#register').removeAttr('disabled').click(registerUsername);
+ $('#registerset').removeAttr('disabled');
+ } else {
+ // Reset status
+ sipcall.hangup();
+ $('#dovideo').removeAttr('disabled').val('');
+ $('#peer').removeAttr('disabled').val('');
+ $('#call').removeAttr('disabled').html('Call')
+ .removeClass("btn-danger").addClass("btn-success")
+ .unbind('click').click(doCall);
+ }
+ bootbox.alert(error);
+ return;
+ }
+ var result = msg["result"];
+ if(result !== null && result !== undefined && result["event"] !== undefined && result["event"] !== null) {
+ var event = result["event"];
+ if(event === 'registration_failed') {
+ Janus.warn("Registration failed: " + result["code"] + " " + result["reason"]);
+ $('#server').removeAttr('disabled');
+ $('#username').removeAttr('disabled');
+ $('#displayname').removeAttr('disabled');
+ $('#password').removeAttr('disabled');
+ $('#register').removeAttr('disabled').click(registerUsername);
+ $('#registerset').removeAttr('disabled');
+ bootbox.alert(result["code"] + " " + result["reason"]);
+ return;
+ }
+ if(event === 'registered') {
+ Janus.log("Successfully registered as " + result["username"] + "!");
+ $('#you').removeClass('hide').show().text("Registered as '" + result["username"] + "'");
+ // TODO Enable buttons to call now
+ if(!registered) {
+ registered = true;
+ $('#phone').removeClass('hide').show();
+ $('#call').unbind('click').click(doCall);
+ $('#peer').focus();
+ }
+ } else if(event === 'calling') {
+ Janus.log("Waiting for the peer to answer...");
+ // TODO Any ringtone?
+ $('#call').removeAttr('disabled').html('Hangup')
+ .removeClass("btn-success").addClass("btn-danger")
+ .unbind('click').click(doHangup);
+ } else if(event === 'incomingcall') {
+ Janus.log("Incoming call from " + result["username"] + "!");
+ var doAudio = true, doVideo = true;
+ if(jsep !== null && jsep !== undefined) {
+ // What has been negotiated?
+ doAudio = (jsep.sdp.indexOf("m=audio ") > -1);
+ doVideo = (jsep.sdp.indexOf("m=video ") > -1);
+ Janus.debug("Audio " + (doAudio ? "has" : "has NOT") + " been negotiated");
+ Janus.debug("Video " + (doVideo ? "has" : "has NOT") + " been negotiated");
+ }
+ // Any security offered? A missing "srtp" attribute means plain RTP
+ var rtpType = "";
+ var srtp = result["srtp"];
+ if(srtp === "sdes_optional")
+ rtpType = " (SDES-SRTP offered)";
+ else if(srtp === "sdes_mandatory")
+ rtpType = " (SDES-SRTP mandatory)";
+ // Notify user
+ bootbox.hideAll();
+ incoming = bootbox.dialog({
+ message: "Incoming call from " + result["username"] + "!" + rtpType,
+ title: "Incoming call",
+ closeButton: false,
+ buttons: {
+ success: {
+ label: "Answer",
+ className: "btn-success",
+ callback: function() {
+ incoming = null;
+ $('#peer').val(result["username"]).attr('disabled', true);
+ sipcall.createAnswer(
+ {
+ jsep: jsep,
+ media: { audio: doAudio, video: doVideo },
+ success: function(jsep) {
+ Janus.debug("Got SDP! audio=" + doAudio + ", video=" + doVideo);
+ Janus.debug(jsep);
+ var body = { request: "accept" };
+ // Note: as with "call", you can add a "srtp" attribute to
+ // negotiate/mandate SDES support for this incoming call.
+ // The default behaviour is to automatically use it if
+ // the caller negotiated it, but you may choose to require
+ // SDES support by setting "srtp" to "sdes_mandatory", e.g.:
+ // var body = { request: "accept", srtp: "sdes_mandatory" };
+ // This way you'll tell the plugin to accept the call, but ONLY
+ // if SDES is available, and you don't want plain RTP. If it
+ // is not available, you'll get an error (452) back.
+ sipcall.send({"message": body, "jsep": jsep});
+ $('#call').removeAttr('disabled').html('Hangup')
+ .removeClass("btn-success").addClass("btn-danger")
+ .unbind('click').click(doHangup);
+ },
+ error: function(error) {
+ Janus.error("WebRTC error:", error);
+ bootbox.alert("WebRTC error... " + JSON.stringify(error));
+ // Don't keep the caller waiting any longer, but use a 480 instead of the default 486 to clarify the cause
+ var body = { "request": "decline", "code": 480 };
+ sipcall.send({"message": body});
+ }
+ });
+ }
+ },
+ danger: {
+ label: "Decline",
+ className: "btn-danger",
+ callback: function() {
+ incoming = null;
+ var body = { "request": "decline" };
+ sipcall.send({"message": body});
+ }
+ }
+ }
+ });
+ } else if(event === 'accepted') {
+ Janus.log(result["username"] + " accepted the call!");
+ // TODO Video call can start
+ if(jsep !== null && jsep !== undefined) {
+ sipcall.handleRemoteJsep({jsep: jsep, error: doHangup });
+ }
+ } else if(event === 'hangup') {
+ if(incoming != null) {
+ incoming.modal('hide');
+ incoming = null;
+ }
+ Janus.log("Call hung up (" + result["code"] + " " + result["reason"] + ")!");
+ bootbox.alert(result["code"] + " " + result["reason"]);
+ // Reset status
+ sipcall.hangup();
+ $('#dovideo').removeAttr('disabled').val('');
+ $('#peer').removeAttr('disabled').val('');
+ $('#call').removeAttr('disabled').html('Call')
+ .removeClass("btn-danger").addClass("btn-success")
+ .unbind('click').click(doCall);
+ }
+ }
+ },
+ onlocalstream: function(stream) {
+ Janus.debug(" ::: Got a local stream :::");
+ Janus.debug(JSON.stringify(stream));
+ $('#videos').removeClass('hide').show();
+ if($('#myvideo').length === 0)
+ $('#videoleft').append('<video class="rounded centered" id="myvideo" width=320 height=240 autoplay muted="muted"/>');
+ Janus.attachMediaStream($('#myvideo').get(0), stream);
+ $("#myvideo").get(0).muted = "muted";
+ // No remote video yet
+ $('#videoright').append('<video class="rounded centered" id="waitingvideo" width=320 height=240 />');
+ if(spinner == null) {
+ var target = document.getElementById('videoright');
+ spinner = new Spinner({top:100}).spin(target);
+ } else {
+ spinner.spin();
+ }
+ var videoTracks = stream.getVideoTracks();
+ if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
+ // No webcam
+ $('#myvideo').hide();
+ $('#videoleft').append(
+ '<div class="no-video-container">' +
+ '<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+ '<span class="no-video-text">No webcam available</span>' +
+ '</div>');
+ }
+ },
+ onremotestream: function(stream) {
+ Janus.debug(" ::: Got a remote stream :::");
+ Janus.debug(JSON.stringify(stream));
+ if($('#remotevideo').length === 0) {
+ $('#videoright').parent().find('h3').html(
+ 'Send DTMF: <span id="dtmf" class="btn-group btn-group-xs"></span>');
+ $('#videoright').append(
+ '<video class="rounded centered hide" id="remotevideo" width=320 height=240 autoplay/>');
+ for(var i=0; i<12; i++) {
+ if(i<10)
+ $('#dtmf').append('<button class="btn btn-info dtmf">' + i + '</button>');
+ else if(i == 10)
+ $('#dtmf').append('<button class="btn btn-info dtmf">#</button>');
+ else if(i == 11)
+ $('#dtmf').append('<button class="btn btn-info dtmf">*</button>');
+ }
+ $('.dtmf').click(function() {
+ if(adapter.browserDetails.browser === 'chrome') {
+ // Send DTMF tone (inband)
+ sipcall.dtmf({dtmf: { tones: $(this).text()}});
+ } else {
+ // Try sending the DTMF tone using SIP INFO
+ sipcall.send({message: {request: "dtmf_info", digit: $(this).text()}});
+ }
+ });
+ }
+ // Show the peer and hide the spinner when we get a playing event
+ $("#remotevideo").bind("playing", function () {
+ $('#waitingvideo').remove();
+ $('#remotevideo').removeClass('hide');
+ if(spinner !== null && spinner !== undefined)
+ spinner.stop();
+ spinner = null;
+ });
+ Janus.attachMediaStream($('#remotevideo').get(0), stream);
+ var videoTracks = stream.getVideoTracks();
+ if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0 || videoTracks[0].muted) {
+ // No remote video
+ $('#remotevideo').hide();
+ $('#videoright').append(
+ '<div class="no-video-container">' +
+ '<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+ '<span class="no-video-text">No remote video available</span>' +
+ '</div>');
+ }
+ },
+ oncleanup: function() {
+ Janus.log(" ::: Got a cleanup notification :::");
+ $('#myvideo').remove();
+ $('#waitingvideo').remove();
+ $('#remotevideo').remove();
+ $('.no-video-container').remove();
+ $('#videos').hide();
+ $('#dtmf').parent().html("Remote UA");
+ }
+ });
+ },
+ error: function(error) {
+ Janus.error(error);
+ bootbox.alert(error, function() {
+ window.location.reload();
+ });
+ },
+ destroyed: function() {
+ window.location.reload();
+ }
+ });
+ });
+ }});
+});
+
+function checkEnter(field, event) {
+ var theCode = event.keyCode ? event.keyCode : event.which ? event.which : event.charCode;
+ if(theCode == 13) {
+ if(field.id == 'server' || field.id == 'username' || field.id == 'password' || field.id == 'displayname')
+ registerUsername();
+ else if(field.id == 'peer')
+ doCall();
+ return false;
+ } else {
+ return true;
+ }
+}
+
+function registerUsername() {
+ if(selectedApproach === null || selectedApproach === undefined) {
+ bootbox.alert("Please select a registration approach from the dropdown menu");
+ return;
+ }
+ // Try a registration
+ $('#server').attr('disabled', true);
+ $('#username').attr('disabled', true);
+ $('#displayname').attr('disabled', true);
+ $('#password').attr('disabled', true);
+ $('#register').attr('disabled', true).unbind('click');
+ $('#registerset').attr('disabled', true);
+ var sipserver = $('#server').val();
+ if(sipserver !== "" && sipserver.indexOf("sip:") != 0 && sipserver.indexOf("sips:") !=0) {
+ bootbox.alert("Please insert a valid SIP server (e.g., sip:192.168.0.1:5060)");
+ $('#server').removeAttr('disabled');
+ $('#username').removeAttr('disabled');
+ $('#displayname').removeAttr('disabled');
+ $('#password').removeAttr('disabled');
+ $('#register').removeAttr('disabled').click(registerUsername);
+ $('#registerset').removeAttr('disabled');
+ return;
+ }
+ if(selectedApproach === "guest") {
+ // We're registering as guests, no username/secret provided
+ var register = {
+ "request" : "register",
+ "type" : "guest"
+ };
+ if(sipserver !== "")
+ register["proxy"] = sipserver;
+ var username = $('#username').val();
+ if(username !== undefined && username !== null) {
+ if(username === "" || username.indexOf("sip:") != 0 || username.indexOf("@") < 0) {
+ bootbox.alert('Usernames are optional for guests: if you want to specify one anyway, though, please insert a valid SIP address (e.g., sip:goofy at example.com)');
+ $('#server').removeAttr('disabled');
+ $('#username').removeAttr('disabled');
+ $('#displayname').removeAttr('disabled');
+ $('#register').removeAttr('disabled').click(registerUsername);
+ $('#registerset').removeAttr('disabled');
+ return;
+ }
+ register.username = username;
+ }
+ var displayname = $('#displayname').val();
+ if (displayname) {
+ register.display_name = displayname;
+ }
+ if(sipserver === "") {
+ bootbox.confirm("You didn't specify a SIP Proxy to use: this will cause the plugin to try and conduct a standard (<a href='https://tools.ietf.org/html/rfc3263' target='_blank'>RFC3263</a>) lookup. If this is not what you want or you don't know what this means, hit Cancel and provide a SIP proxy instead'",
+ function(result) {
+ if(result) {
+ sipcall.send({"message": register});
+ } else {
+ $('#server').removeAttr('disabled');
+ $('#username').removeAttr('disabled');
+ $('#displayname').removeAttr('disabled');
+ $('#register').removeAttr('disabled').click(registerUsername);
+ $('#registerset').removeAttr('disabled');
+ }
+ });
+ } else {
+ sipcall.send({"message": register});
+ }
+ return;
+ }
+ var username = $('#username').val();
+ if(username === "" || username.indexOf("sip:") != 0 || username.indexOf("@") < 0) {
+ bootbox.alert('Please insert a valid SIP identity address (e.g., sip:goofy at example.com)');
+ $('#server').removeAttr('disabled');
+ $('#username').removeAttr('disabled');
+ $('#displayname').removeAttr('disabled');
+ $('#password').removeAttr('disabled');
+ $('#register').removeAttr('disabled').click(registerUsername);
+ $('#registerset').removeAttr('disabled');
+ return;
+ }
+ var password = $('#password').val();
+ if(password === "") {
+ bootbox.alert("Insert the username secret (e.g., mypassword)");
+ $('#server').removeAttr('disabled');
+ $('#username').removeAttr('disabled');
+ $('#displayname').removeAttr('disabled');
+ $('#password').removeAttr('disabled');
+ $('#register').removeAttr('disabled').click(registerUsername);
+ $('#registerset').removeAttr('disabled');
+ return;
+ }
+ var register = {
+ "request" : "register",
+ "username" : username
+ };
+ var displayname = $('#displayname').val();
+ if (displayname) {
+ register.display_name = displayname;
+ }
+ if(selectedApproach === "secret") {
+ // Use the plain secret
+ register["secret"] = password;
+ } else if(selectedApproach === "ha1secret") {
+ var sip_user = username.substring(4, username.indexOf('@')); /* skip sip: */
+ var sip_domain = username.substring(username.indexOf('@')+1);
+ register["ha1_secret"] = md5(sip_user+':'+sip_domain+':'+password);
+ }
+ if(sipserver === "") {
+ bootbox.confirm("You didn't specify a SIP Proxy to use: this will cause the plugin to try and conduct a standard (<a href='https://tools.ietf.org/html/rfc3263' target='_blank'>RFC3263</a>) lookup. If this is not what you want or you don't know what this means, hit Cancel and provide a SIP proxy instead'",
+ function(result) {
+ if(result) {
+ sipcall.send({"message": register});
+ } else {
+ $('#server').removeAttr('disabled');
+ $('#username').removeAttr('disabled');
+ $('#displayname').removeAttr('disabled');
+ $('#password').removeAttr('disabled');
+ $('#register').removeAttr('disabled').click(registerUsername);
+ $('#registerset').removeAttr('disabled');
+ }
+ });
+ } else {
+ register["proxy"] = sipserver;
+ sipcall.send({"message": register});
+ }
+}
+
+function doCall() {
+ // Call someone
+ $('#peer').attr('disabled', true);
+ $('#call').attr('disabled', true).unbind('click');
+ $('#dovideo').attr('disabled', true);
+ var username = $('#peer').val();
+ if(username === "") {
+ bootbox.alert('Please insert a valid SIP address (e.g., sip:pluto at example.com)');
+ $('#peer').removeAttr('disabled');
+ $('#dovideo').removeAttr('disabled');
+ $('#call').removeAttr('disabled').click(doCall);
+ return;
+ }
+ if(username.indexOf("sip:") != 0 || username.indexOf("@") < 0) {
+ bootbox.alert('Please insert a valid SIP address (e.g., sip:pluto at example.com)');
+ $('#peer').removeAttr('disabled').val("");
+ $('#dovideo').removeAttr('disabled').val("");
+ $('#call').removeAttr('disabled').click(doCall);
+ return;
+ }
+ // Call this URI
+ doVideo = $('#dovideo').is(':checked');
+ Janus.log("This is a SIP " + (doVideo ? "video" : "audio") + " call (dovideo=" + doVideo + ")");
+ sipcall.createOffer(
+ {
+ media: {
+ audioSend: true, audioRecv: true, // We DO want audio
+ videoSend: doVideo, videoRecv: doVideo // We MAY want video
+ },
+ success: function(jsep) {
+ Janus.debug("Got SDP!");
+ Janus.debug(jsep);
+ // By default, you only pass the SIP URI to call as an
+ // argument to a "call" request. Should you want the
+ // SIP stack to add some custom headers to the INVITE,
+ // you can do so by adding an additional "headers" object,
+ // containing each of the headers as key-value, e.g.:
+ // var body = { request: "call", uri: $('#peer').val(),
+ // headers: {
+ // "My-Header": "value",
+ // "AnotherHeader": "another string"
+ // }
+ // };
+ var body = { request: "call", uri: $('#peer').val() };
+ // Note: you can also ask the plugin to negotiate SDES-SRTP, instead of the
+ // default plain RTP, by adding a "srtp" attribute to the request. Valid
+ // values are "sdes_optional" and "sdes_mandatory", e.g.:
+ // var body = { request: "call", uri: $('#peer').val(), srtp: "sdes_optional" };
+ // "sdes_optional" will negotiate RTP/AVP and add a crypto line,
+ // "sdes_mandatory" will set the protocol to RTP/SAVP instead.
+ // Just beware that some endpoints will NOT accept an INVITE
+ // with a crypto line in it if the protocol is not RTP/SAVP,
+ // so if you want SDES use "sdes_optional" with care.
+ sipcall.send({"message": body, "jsep": jsep});
+ },
+ error: function(error) {
+ Janus.error("WebRTC error...", error);
+ bootbox.alert("WebRTC error... " + JSON.stringify(error));
+ }
+ });
+}
+
+function doHangup() {
+ // Hangup a call
+ $('#call').attr('disabled', true).unbind('click');
+ var hangup = { "request": "hangup" };
+ sipcall.send({"message": hangup});
+ sipcall.hangup();
+}
diff --git a/html/siptest.html b/html/siptest.html
index d1dcfd8..fe21481 100644
--- a/html/siptest.html
+++ b/html/siptest.html
@@ -4,7 +4,7 @@
<meta charset="utf-8">
<meta name="viewport" content="width=device-width, initial-scale=1.0"/>
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8" />
-<title>Janus WebRTC Gateway: SIP Gateway Demo</title>
+<title>Janus WebRTC Gateway: SIP Gateway Demo (Sofia)</title>
<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/webrtc-adapter/3.1.5/adapter.min.js" ></script>
<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/jquery/1.7.2/jquery.min.js" ></script>
<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/jquery.blockUI/2.70/jquery.blockUI.min.js" ></script>
@@ -38,7 +38,7 @@
<div class="row">
<div class="col-md-12">
<div class="page-header">
- <h1>Plugin Demo: SIP Gateway
+ <h1>Plugin Demo: SIP Gateway (Sofia)
<button class="btn btn-default" autocomplete="off" id="start">Start</button>
</h1>
</div>
@@ -48,7 +48,11 @@
<h3>Demo details</h3>
<p>This demo shows how you can make use of the SIP plugin to interact with a
SIP Proxy (e.g., Kamailio) or PBX (e.g., Asterisk) in order to place or
- receive calls to and from other SIP clients.</p>
+ receive calls to and from other SIP clients. Specifically, it uses the Sofia-based
+ SIP plugin: in case you're interested in the libre-based one, check
+ <a href="sipretest.html">this other demo</a> instead. Notice that both
+ plugins only exchange SIP messages from within the plugin itself: no SIP
+ is done in JavaScript, except for references to SIP URIs.</p>
<p>When started, the demo will allow you to insert a minimum set of information
required to REGISTER the web page as a SIP client at a SIP Proxy or PBX you specify.
This will allow you to call SIP URIs, or receive calls through the SIP Server itself.
diff --git a/plugins/janus_sipre.c b/plugins/janus_sipre.c
new file mode 100644
index 0000000..4499def
--- /dev/null
+++ b/plugins/janus_sipre.c
@@ -0,0 +1,2937 @@
+/*! \file janus_sipre.c
+ * \author Lorenzo Miniero <lorenzo at meetecho.com>
+ * \copyright GNU General Public License v3
+ * \brief Janus SIPre plugin (libre)
+ * \details This is basically a clone of the SIPre plugin, with the key
+ * difference being that it uses \c libre (http://creytiv.com/re.html)
+ * instead of Sofia SIP for its internal stack. As such, it provides an
+ * alternative for those who don't want to, or can't, use the Sofia-based
+ * SIP plugin. The API it exposes is exactly the same, meaning it should
+ * be pretty straightforward to switch from one plugin to another on the
+ * client side. The configuration file looks exactly the same as well.
+ *
+ * \section sipapi SIPre Plugin API
+ *
+ * All requests you can send in the SIPre Plugin API are asynchronous,
+ * which means all responses (successes and errors) will be delivered
+ * as events with the same transaction.
+ *
+ * The supported requests are \c register , \c call , \c accept and
+ * \c hangup . \c register can be used, as the name suggests, to register
+ * a username at a SIPre registrar to call and be called; \c call is used
+ * to send an INVITE to a different SIPre URI through the plugin, while
+ * \c accept is used to accept the call in case one is invited instead
+ * of inviting; finally, \c hangup can be used to terminate the
+ * communication at any time, either to hangup (BYE) an ongoing call or
+ * to cancel/decline (CANCEL/BYE) a call that hasn't started yet.
+ *
+ * Actual API docs: TBD.
+ *
+ * \ingroup plugins
+ * \ref plugins
+ */
+
+#include "plugin.h"
+
+#include <arpa/inet.h>
+#include <net/if.h>
+#include <sys/socket.h>
+#include <netdb.h>
+#include <poll.h>
+
+#include <jansson.h>
+
+#include <re_types.h>
+#include <re_fmt.h>
+#include <re_mbuf.h>
+#include <re_msg.h>
+#include <re_list.h>
+#include <re_sa.h>
+#include <re_main.h>
+#include <re_mem.h>
+#include <re_sdp.h>
+#include <re_uri.h>
+#include <re_sip.h>
+#include <re_sipreg.h>
+#include <re_sipsess.h>
+#include <re_srtp.h>
+#include <re_tmr.h>
+#include <re_tls.h>
+
+#include "../debug.h"
+#include "../apierror.h"
+#include "../config.h"
+#include "../mutex.h"
+#include "../record.h"
+#include "../rtp.h"
+#include "../rtcp.h"
+#include "../sdp-utils.h"
+#include "../utils.h"
+
+
+/* Plugin information */
+#define JANUS_SIPRE_VERSION 1
+#define JANUS_SIPRE_VERSION_STRING "0.0.1"
+#define JANUS_SIPRE_DESCRIPTION "This is a simple SIP plugin for Janus (based on libre instead of Sofia), allowing WebRTC peers to register at a SIP server and call SIP user agents through the gateway."
+#define JANUS_SIPRE_NAME "JANUS SIPre plugin"
+#define JANUS_SIPRE_AUTHOR "Meetecho s.r.l."
+#define JANUS_SIPRE_PACKAGE "janus.plugin.sipre"
+
+/* Plugin methods */
+janus_plugin *create(void);
+int janus_sipre_init(janus_callbacks *callback, const char *config_path);
+void janus_sipre_destroy(void);
+int janus_sipre_get_api_compatibility(void);
+int janus_sipre_get_version(void);
+const char *janus_sipre_get_version_string(void);
+const char *janus_sipre_get_description(void);
+const char *janus_sipre_get_name(void);
+const char *janus_sipre_get_author(void);
+const char *janus_sipre_get_package(void);
+void janus_sipre_create_session(janus_plugin_session *handle, int *error);
+struct janus_plugin_result *janus_sipre_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
+void janus_sipre_setup_media(janus_plugin_session *handle);
+void janus_sipre_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_sipre_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_sipre_hangup_media(janus_plugin_session *handle);
+void janus_sipre_destroy_session(janus_plugin_session *handle, int *error);
+json_t *janus_sipre_query_session(janus_plugin_session *handle);
+
+/* Plugin setup */
+static janus_plugin janus_sipre_plugin =
+ JANUS_PLUGIN_INIT (
+ .init = janus_sipre_init,
+ .destroy = janus_sipre_destroy,
+
+ .get_api_compatibility = janus_sipre_get_api_compatibility,
+ .get_version = janus_sipre_get_version,
+ .get_version_string = janus_sipre_get_version_string,
+ .get_description = janus_sipre_get_description,
+ .get_name = janus_sipre_get_name,
+ .get_author = janus_sipre_get_author,
+ .get_package = janus_sipre_get_package,
+
+ .create_session = janus_sipre_create_session,
+ .handle_message = janus_sipre_handle_message,
+ .setup_media = janus_sipre_setup_media,
+ .incoming_rtp = janus_sipre_incoming_rtp,
+ .incoming_rtcp = janus_sipre_incoming_rtcp,
+ .hangup_media = janus_sipre_hangup_media,
+ .destroy_session = janus_sipre_destroy_session,
+ .query_session = janus_sipre_query_session,
+ );
+
+/* Plugin creator */
+janus_plugin *create(void) {
+ JANUS_LOG(LOG_VERB, "%s created!\n", JANUS_SIPRE_NAME);
+ return &janus_sipre_plugin;
+}
+
+/* Parameter validation */
+static struct janus_json_parameter request_parameters[] = {
+ {"request", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+};
+static struct janus_json_parameter register_parameters[] = {
+ {"type", JANUS_JSON_STRING, 0},
+ {"send_register", JANUS_JSON_BOOL, 0},
+ {"sips", JANUS_JSON_BOOL, 0},
+ {"username", JANUS_JSON_STRING, 0},
+ {"secret", JANUS_JSON_STRING, 0},
+ {"ha1_secret", JANUS_JSON_STRING, 0},
+ {"authuser", JANUS_JSON_STRING, 0}
+};
+static struct janus_json_parameter proxy_parameters[] = {
+ {"proxy", JANUS_JSON_STRING, 0}
+};
+static struct janus_json_parameter call_parameters[] = {
+ {"uri", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+ {"autoack", JANUS_JSON_BOOL, 0},
+ {"headers", JANUS_JSON_OBJECT, 0},
+ {"srtp", JANUS_JSON_STRING, 0}
+};
+static struct janus_json_parameter accept_parameters[] = {
+ {"srtp", JANUS_JSON_STRING, 0}
+};
+static struct janus_json_parameter recording_parameters[] = {
+ {"action", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+ {"audio", JANUS_JSON_BOOL, 0},
+ {"video", JANUS_JSON_BOOL, 0},
+ {"peer_audio", JANUS_JSON_BOOL, 0},
+ {"peer_video", JANUS_JSON_BOOL, 0},
+ {"filename", JANUS_JSON_STRING, 0}
+};
+static struct janus_json_parameter dtmf_info_parameters[] = {
+ {"digit", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+ {"duration", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
+};
+
+/* Useful stuff */
+static volatile gint initialized = 0, stopping = 0;
+static gboolean notify_events = TRUE;
+static janus_callbacks *gateway = NULL;
+
+static char local_ip[INET6_ADDRSTRLEN];
+static int keepalive_interval = 120;
+static gboolean behind_nat = FALSE;
+static char *user_agent;
+#define JANUS_DEFAULT_REGISTER_TTL 3600
+static int register_ttl = JANUS_DEFAULT_REGISTER_TTL;
+
+static GThread *handler_thread;
+static GThread *watchdog;
+static void *janus_sipre_handler(void *data);
+
+typedef struct janus_sipre_message {
+ janus_plugin_session *handle;
+ char *transaction;
+ json_t *message;
+ json_t *jsep;
+} janus_sipre_message;
+static GAsyncQueue *messages = NULL;
+static janus_sipre_message exit_message;
+
+static void janus_sipre_message_free(janus_sipre_message *msg) {
+ if(!msg || msg == &exit_message)
+ return;
+
+ msg->handle = NULL;
+
+ g_free(msg->transaction);
+ msg->transaction = NULL;
+ if(msg->message)
+ json_decref(msg->message);
+ msg->message = NULL;
+ if(msg->jsep)
+ json_decref(msg->jsep);
+ msg->jsep = NULL;
+
+ g_free(msg);
+}
+
+/* libre SIP stack */
+static struct sip *sipstack;
+static struct tls *tls = NULL;
+GThread *sipstack_thread = NULL;
+
+/* Registration info */
+typedef enum {
+ janus_sipre_registration_status_disabled = -2,
+ janus_sipre_registration_status_failed = -1,
+ janus_sipre_registration_status_unregistered = 0,
+ janus_sipre_registration_status_registering,
+ janus_sipre_registration_status_registered,
+ janus_sipre_registration_status_unregistering,
+} janus_sipre_registration_status;
+
+static const char *janus_sipre_registration_status_string(janus_sipre_registration_status status) {
+ switch(status) {
+ case janus_sipre_registration_status_disabled:
+ return "disabled";
+ case janus_sipre_registration_status_failed:
+ return "failed";
+ case janus_sipre_registration_status_unregistered:
+ return "unregistered";
+ case janus_sipre_registration_status_registering:
+ return "registering";
+ case janus_sipre_registration_status_registered:
+ return "registered";
+ case janus_sipre_registration_status_unregistering:
+ return "unregistering";
+ default:
+ return "unknown";
+ }
+}
+
+
+typedef enum {
+ janus_sipre_call_status_idle = 0,
+ janus_sipre_call_status_inviting,
+ janus_sipre_call_status_invited,
+ janus_sipre_call_status_incall,
+ janus_sipre_call_status_closing,
+} janus_sipre_call_status;
+
+static const char *janus_sipre_call_status_string(janus_sipre_call_status status) {
+ switch(status) {
+ case janus_sipre_call_status_idle:
+ return "idle";
+ case janus_sipre_call_status_inviting:
+ return "inviting";
+ case janus_sipre_call_status_invited:
+ return "invited";
+ case janus_sipre_call_status_incall:
+ return "incall";
+ case janus_sipre_call_status_closing:
+ return "closing";
+ default:
+ return "unknown";
+ }
+}
+
+
+typedef enum {
+ janus_sipre_secret_type_plaintext = 1,
+ janus_sipre_secret_type_hashed = 2,
+ janus_sipre_secret_type_unknown
+} janus_sipre_secret_type;
+
+typedef struct janus_sipre_account {
+ char *identity;
+ char *user_agent; /* Used to override the general UA string */
+ gboolean sips;
+ char *username;
+ char *display_name; /* Used for outgoing calls in the From header */
+ char *authuser; /**< username to use for authentication */
+ char *secret;
+ janus_sipre_secret_type secret_type;
+ int sip_port;
+ char *proxy;
+ janus_sipre_registration_status registration_status;
+} janus_sipre_account;
+
+typedef struct janus_sipre_stack {
+ struct sipsess *sess; /* SIP session */
+ struct sipsess_sock *sess_sock; /* SIP session socket */
+ struct sipreg *reg; /* SIP registration */
+ struct sdp_session *sdp; /* SDP session */
+ struct sdp_media *sdp_media; /* SDP media */
+ void *session; /* Opaque pointer to the plugin session */
+} janus_sipre_stack;
+
+typedef struct janus_sipre_media {
+ char *remote_ip;
+ int ready:1;
+ gboolean autoack;
+ gboolean require_srtp, has_srtp_local, has_srtp_remote;
+ int has_audio:1;
+ int audio_rtp_fd, audio_rtcp_fd;
+ int local_audio_rtp_port, remote_audio_rtp_port;
+ int local_audio_rtcp_port, remote_audio_rtcp_port;
+ guint32 audio_ssrc, audio_ssrc_peer;
+ int audio_pt;
+ const char *audio_pt_name;
+ srtp_t audio_srtp_in, audio_srtp_out;
+ srtp_policy_t audio_remote_policy, audio_local_policy;
+ int audio_srtp_suite_in, audio_srtp_suite_out;
+ gboolean audio_send;
+ int has_video:1;
+ int video_rtp_fd, video_rtcp_fd;
+ int local_video_rtp_port, remote_video_rtp_port;
+ int local_video_rtcp_port, remote_video_rtcp_port;
+ guint32 video_ssrc, video_ssrc_peer;
+ int video_pt;
+ const char *video_pt_name;
+ srtp_t video_srtp_in, video_srtp_out;
+ srtp_policy_t video_remote_policy, video_local_policy;
+ int video_srtp_suite_in, video_srtp_suite_out;
+ gboolean video_send;
+ janus_rtp_switching_context context;
+ int pipefd[2];
+ gboolean updated;
+} janus_sipre_media;
+
+typedef struct janus_sipre_session {
+ janus_plugin_session *handle;
+ janus_sipre_stack stack;
+ janus_sipre_account account;
+ janus_sipre_call_status status;
+ janus_sipre_media media;
+ char *transaction;
+ char *callee;
+ char *callid;
+ janus_sdp *sdp; /* The SDP this user sent */
+ janus_recorder *arc; /* The Janus recorder instance for this user's audio, if enabled */
+ janus_recorder *arc_peer; /* The Janus recorder instance for the peer's audio, if enabled */
+ janus_recorder *vrc; /* The Janus recorder instance for this user's video, if enabled */
+ janus_recorder *vrc_peer; /* The Janus recorder instance for the peer's video, if enabled */
+ janus_mutex rec_mutex; /* Mutex to protect the recorders from race conditions */
+ volatile gint hangingup;
+ gint64 destroyed; /* Time at which this session was marked as destroyed */
+ janus_mutex mutex;
+} janus_sipre_session;
+static GHashTable *sessions;
+static GList *old_sessions;
+static GHashTable *identities;
+static GHashTable *callids;
+static janus_mutex sessions_mutex;
+
+
+/* SRTP stuff (in case we need SDES) */
+static int janus_sipre_srtp_set_local(janus_sipre_session *session, gboolean video, char **crypto) {
+ if(session == NULL)
+ return -1;
+ /* Generate key/salt */
+ uint8_t *key = g_malloc0(SRTP_MASTER_LENGTH);
+ srtp_crypto_get_random(key, SRTP_MASTER_LENGTH);
+ /* Set SRTP policies */
+ srtp_policy_t *policy = video ? &session->media.video_local_policy : &session->media.audio_local_policy;
+ srtp_crypto_policy_set_rtp_default(&(policy->rtp));
+ srtp_crypto_policy_set_rtcp_default(&(policy->rtcp));
+ policy->ssrc.type = ssrc_any_inbound;
+ policy->key = key;
+ policy->next = NULL;
+ /* Create SRTP context */
+ srtp_err_status_t res = srtp_create(video ? &session->media.video_srtp_out : &session->media.audio_srtp_out, policy);
+ if(res != srtp_err_status_ok) {
+ /* Something went wrong... */
+ JANUS_LOG(LOG_ERR, "Oops, error creating outbound SRTP session: %d (%s)\n", res, janus_srtp_error_str(res));
+ g_free(key);
+ policy->key = NULL;
+ return -2;
+ }
+ /* Base64 encode the salt */
+ *crypto = g_base64_encode(key, SRTP_MASTER_LENGTH);
+ if((video && session->media.video_srtp_out) || (!video && session->media.audio_srtp_out)) {
+ JANUS_LOG(LOG_VERB, "%s outbound SRTP session created\n", video ? "Video" : "Audio");
+ }
+ return 0;
+}
+static int janus_sipre_srtp_set_remote(janus_sipre_session *session, gboolean video, const char *crypto, int suite) {
+ if(session == NULL || crypto == NULL)
+ return -1;
+ /* Base64 decode the crypto string and set it as the remote SRTP context */
+ gsize len = 0;
+ guchar *decoded = g_base64_decode(crypto, &len);
+ if(len < SRTP_MASTER_LENGTH) {
+ /* FIXME Can this happen? */
+ g_free(decoded);
+ return -2;
+ }
+ /* Set SRTP policies */
+ srtp_policy_t *policy = video ? &session->media.video_remote_policy : &session->media.audio_remote_policy;
+ srtp_crypto_policy_set_rtp_default(&(policy->rtp));
+ srtp_crypto_policy_set_rtcp_default(&(policy->rtcp));
+ if(suite == 32) {
+ srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&(policy->rtp));
+ srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&(policy->rtcp));
+ } else if(suite == 80) {
+ srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&(policy->rtp));
+ srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&(policy->rtcp));
+ }
+ policy->ssrc.type = ssrc_any_inbound;
+ policy->key = decoded;
+ policy->next = NULL;
+ /* Create SRTP context */
+ srtp_err_status_t res = srtp_create(video ? &session->media.video_srtp_in : &session->media.audio_srtp_in, policy);
+ if(res != srtp_err_status_ok) {
+ /* Something went wrong... */
+ JANUS_LOG(LOG_ERR, "Oops, error creating inbound SRTP session: %d (%s)\n", res, janus_srtp_error_str(res));
+ g_free(decoded);
+ policy->key = NULL;
+ return -2;
+ }
+ if((video && session->media.video_srtp_in) || (!video && session->media.audio_srtp_in)) {
+ JANUS_LOG(LOG_VERB, "%s inbound SRTP session created\n", video ? "Video" : "Audio");
+ }
+ return 0;
+}
+static void janus_sipre_srtp_cleanup(janus_sipre_session *session) {
+ if(session == NULL)
+ return;
+ session->media.autoack = TRUE;
+ session->media.require_srtp = FALSE;
+ session->media.has_srtp_local = FALSE;
+ session->media.has_srtp_remote = FALSE;
+ /* Audio */
+ if(session->media.audio_srtp_out)
+ srtp_dealloc(session->media.audio_srtp_out);
+ session->media.audio_srtp_out = NULL;
+ g_free(session->media.audio_local_policy.key);
+ session->media.audio_local_policy.key = NULL;
+ session->media.audio_srtp_suite_out = 0;
+ if(session->media.audio_srtp_in)
+ srtp_dealloc(session->media.audio_srtp_in);
+ session->media.audio_srtp_in = NULL;
+ g_free(session->media.audio_remote_policy.key);
+ session->media.audio_remote_policy.key = NULL;
+ session->media.audio_srtp_suite_in = 0;
+ /* Video */
+ if(session->media.video_srtp_out)
+ srtp_dealloc(session->media.video_srtp_out);
+ session->media.video_srtp_out = NULL;
+ g_free(session->media.video_local_policy.key);
+ session->media.video_local_policy.key = NULL;
+ session->media.video_srtp_suite_out = 0;
+ if(session->media.video_srtp_in)
+ srtp_dealloc(session->media.video_srtp_in);
+ session->media.video_srtp_in = NULL;
+ g_free(session->media.video_remote_policy.key);
+ session->media.video_remote_policy.key = NULL;
+ session->media.video_srtp_suite_in = 0;
+}
+
+
+/* libre event thread */
+gpointer janus_sipre_stack_thread(gpointer user_data);
+/* libre callbacks */
+int janus_sipre_cb_auth(char **user, char **pass, const char *realm, void *arg);
+void janus_sipre_cb_register(int err, const struct sip_msg *msg, void *arg);
+void janus_sipre_cb_progress(const struct sip_msg *msg, void *arg);
+void janus_sipre_cb_incoming(const struct sip_msg *msg, void *arg);
+int janus_sipre_cb_offer(struct mbuf **mbp, const struct sip_msg *msg, void *arg);
+int janus_sipre_cb_answer(const struct sip_msg *msg, void *arg);
+void janus_sipre_cb_established(const struct sip_msg *msg, void *arg);
+void janus_sipre_cb_closed(int err, const struct sip_msg *msg, void *arg);
+void janus_sipre_cb_exit(void *arg);
+
+/* URI parsing utilities */
+static int janus_sipre_parse_uri(const char *uri) {
+ if(uri == NULL)
+ return -1;
+ struct sip_addr addr;
+ struct pl pluri;
+ pl_set_str(&pluri, uri);
+ if(sip_addr_decode(&addr, &pluri) != 0)
+ return -1;
+ return 0;
+}
+static char *janus_sipre_get_uri_username(const char *uri) {
+ if(uri == NULL)
+ return NULL;
+ struct sip_addr addr;
+ struct pl pluri;
+ pl_set_str(&pluri, uri);
+ if(sip_addr_decode(&addr, &pluri) != 0)
+ return NULL;
+ char *at = strchr(addr.uri.user.p, '@');
+ if(at != NULL)
+ *(at) = '\0';
+ char *username = g_strdup(addr.uri.user.p);
+ if(at != NULL)
+ *(at) = '@';
+ return username;
+}
+static char *janus_sipre_get_uri_host(const char *uri) {
+ if(uri == NULL)
+ return NULL;
+ struct sip_addr addr;
+ struct pl pluri;
+ pl_set_str(&pluri, uri);
+ if(sip_addr_decode(&addr, &pluri) != 0)
+ return NULL;
+ return g_strdup(addr.uri.host.p);
+}
+static uint16_t janus_sipre_get_uri_port(const char *uri) {
+ if(uri == NULL)
+ return 0;
+ struct sip_addr addr;
+ struct pl pluri;
+ pl_set_str(&pluri, uri);
+ if(sip_addr_decode(&addr, &pluri) != 0)
+ return 0;
+ return addr.uri.port;
+}
+
+
+/* SDP parsing and manipulation */
+void janus_sipre_sdp_process(janus_sipre_session *session, janus_sdp *sdp, gboolean answer, gboolean update, gboolean *changed);
+char *janus_sipre_sdp_manipulate(janus_sipre_session *session, janus_sdp *sdp, gboolean answer);
+/* Media */
+static int janus_sipre_allocate_local_ports(janus_sipre_session *session);
+static void *janus_sipre_relay_thread(void *data);
+
+
+/* Error codes */
+#define JANUS_SIPRE_ERROR_UNKNOWN_ERROR 499
+#define JANUS_SIPRE_ERROR_NO_MESSAGE 440
+#define JANUS_SIPRE_ERROR_INVALID_JSON 441
+#define JANUS_SIPRE_ERROR_INVALID_REQUEST 442
+#define JANUS_SIPRE_ERROR_MISSING_ELEMENT 443
+#define JANUS_SIPRE_ERROR_INVALID_ELEMENT 444
+#define JANUS_SIPRE_ERROR_ALREADY_REGISTERED 445
+#define JANUS_SIPRE_ERROR_INVALID_ADDRESS 446
+#define JANUS_SIPRE_ERROR_WRONG_STATE 447
+#define JANUS_SIPRE_ERROR_MISSING_SDP 448
+#define JANUS_SIPRE_ERROR_LIBRE_ERROR 449
+#define JANUS_SIPRE_ERROR_IO_ERROR 450
+#define JANUS_SIPRE_ERROR_RECORDING_ERROR 451
+#define JANUS_SIPRE_ERROR_TOO_STRICT 452
+
+
+/* SIPre watchdog/garbage collector (sort of) */
+static void *janus_sipre_watchdog(void *data) {
+ JANUS_LOG(LOG_INFO, "SIPre watchdog started\n");
+ gint64 now = 0;
+ while(g_atomic_int_get(&initialized) && !g_atomic_int_get(&stopping)) {
+ janus_mutex_lock(&sessions_mutex);
+ /* Iterate on all the sessions */
+ now = janus_get_monotonic_time();
+ if(old_sessions != NULL) {
+ GList *sl = old_sessions;
+ JANUS_LOG(LOG_HUGE, "Checking %d old SIPre sessions...\n", g_list_length(old_sessions));
+ while(sl) {
+ janus_sipre_session *session = (janus_sipre_session *)sl->data;
+ if(!session) {
+ sl = sl->next;
+ continue;
+ }
+ if(now-session->destroyed >= 5*G_USEC_PER_SEC) {
+ /* We're lazy and actually get rid of the stuff only after a few seconds */
+ JANUS_LOG(LOG_VERB, "Freeing old SIPre session\n");
+ GList *rm = sl->next;
+ old_sessions = g_list_delete_link(old_sessions, sl);
+ sl = rm;
+ if(session->account.identity) {
+ g_hash_table_remove(identities, session->account.identity);
+ g_free(session->account.identity);
+ session->account.identity = NULL;
+ }
+ session->account.sips = TRUE;
+ if(session->account.proxy) {
+ g_free(session->account.proxy);
+ session->account.proxy = NULL;
+ }
+ if(session->account.secret) {
+ g_free(session->account.secret);
+ session->account.secret = NULL;
+ }
+ if(session->account.username) {
+ g_free(session->account.username);
+ session->account.username = NULL;
+ }
+ if(session->account.display_name) {
+ g_free(session->account.display_name);
+ session->account.display_name = NULL;
+ }
+ if(session->account.user_agent) {
+ g_free(session->account.user_agent);
+ session->account.user_agent = NULL;
+ }
+ if(session->account.authuser) {
+ g_free(session->account.authuser);
+ session->account.authuser = NULL;
+ }
+ if(session->callee) {
+ g_free(session->callee);
+ session->callee = NULL;
+ }
+ if(session->callid) {
+ g_hash_table_remove(callids, session->callid);
+ g_free(session->callid);
+ session->callid = NULL;
+ }
+ if(session->sdp) {
+ janus_sdp_free(session->sdp);
+ session->sdp = NULL;
+ }
+ if(session->transaction) {
+ g_free(session->transaction);
+ session->transaction = NULL;
+ }
+ if(session->media.remote_ip) {
+ g_free(session->media.remote_ip);
+ session->media.remote_ip = NULL;
+ }
+ janus_sipre_srtp_cleanup(session);
+ session->handle = NULL;
+ g_free(session);
+ session = NULL;
+ continue;
+ }
+ sl = sl->next;
+ }
+ }
+ janus_mutex_unlock(&sessions_mutex);
+ g_usleep(500000);
+ }
+ JANUS_LOG(LOG_INFO, "SIPre watchdog stopped\n");
+ return NULL;
+}
+
+
+static void janus_sipre_detect_local_ip(char *buf, size_t buflen) {
+ JANUS_LOG(LOG_VERB, "Autodetecting local IP...\n");
+
+ struct sockaddr_in addr;
+ socklen_t len;
+ int fd = socket(AF_INET, SOCK_DGRAM, 0);
+ if(fd == -1)
+ goto error;
+ addr.sin_family = AF_INET;
+ addr.sin_port = htons(1);
+ inet_pton(AF_INET, "1.2.3.4", &addr.sin_addr.s_addr);
+ if(connect(fd, (const struct sockaddr*) &addr, sizeof(addr)) < 0)
+ goto error;
+ len = sizeof(addr);
+ if(getsockname(fd, (struct sockaddr*) &addr, &len) < 0)
+ goto error;
+ if(getnameinfo((const struct sockaddr*) &addr, sizeof(addr),
+ buf, buflen,
+ NULL, 0, NI_NUMERICHOST) != 0)
+ goto error;
+ close(fd);
+ return;
+
+error:
+ if(fd != -1)
+ close(fd);
+ JANUS_LOG(LOG_VERB, "Couldn't find any address! using 127.0.0.1 as the local IP... (which is NOT going to work out of your machine)\n");
+ g_strlcpy(buf, "127.0.0.1", buflen);
+}
+
+
+/* Random string helper (for call-ids) */
+static char charset[] = "abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ0123456789";
+static void janus_sipre_random_string(int length, char *buffer) {
+ if(length > 0 && buffer) {
+ int l = (int)(sizeof(charset)-1);
+ int i=0;
+ for(i=0; i<length; i++) {
+ int key = rand() % l;
+ buffer[i] = charset[key];
+ }
+ buffer[length-1] = '\0';
+ }
+}
+
+
+/* Plugin implementation */
+int janus_sipre_init(janus_callbacks *callback, const char *config_path) {
+ if(g_atomic_int_get(&stopping)) {
+ /* Still stopping from before */
+ return -1;
+ }
+ if(callback == NULL || config_path == NULL) {
+ /* Invalid arguments */
+ return -1;
+ }
+
+ /* Read configuration */
+ char filename[255];
+ g_snprintf(filename, 255, "%s/%s.cfg", config_path, JANUS_SIPRE_PACKAGE);
+ JANUS_LOG(LOG_VERB, "Configuration file: %s\n", filename);
+ janus_config *config = janus_config_parse(filename);
+ gboolean local_ip_set = FALSE;
+ if(config != NULL) {
+ janus_config_print(config);
+
+ janus_config_item *item = janus_config_get_item_drilldown(config, "general", "local_ip");
+ if(item && item->value) {
+ int family;
+ if(!janus_is_ip_valid(item->value, &family)) {
+ JANUS_LOG(LOG_WARN, "Invalid local IP specified: %s, guessing the default...\n", item->value);
+ } else {
+ /* Verify that we can actually bind to that address */
+ int fd = socket(family, SOCK_DGRAM, 0);
+ if(fd == -1) {
+ JANUS_LOG(LOG_WARN, "Error creating test socket, falling back to detecting IP address...\n");
+ } else {
+ int r;
+ struct sockaddr_storage ss;
+ socklen_t addrlen;
+ memset(&ss, 0, sizeof(ss));
+ if(family == AF_INET) {
+ struct sockaddr_in *addr4 = (struct sockaddr_in*)&ss;
+ addr4->sin_family = AF_INET;
+ addr4->sin_port = 0;
+ inet_pton(AF_INET, item->value, &(addr4->sin_addr.s_addr));
+ addrlen = sizeof(struct sockaddr_in);
+ } else {
+ struct sockaddr_in6 *addr6 = (struct sockaddr_in6*)&ss;
+ addr6->sin6_family = AF_INET6;
+ addr6->sin6_port = 0;
+ inet_pton(AF_INET6, item->value, &(addr6->sin6_addr.s6_addr));
+ addrlen = sizeof(struct sockaddr_in6);
+ }
+ r = bind(fd, (const struct sockaddr*)&ss, addrlen);
+ close(fd);
+ if(r < 0) {
+ JANUS_LOG(LOG_WARN, "Error setting local IP address to %s, falling back to detecting IP address...\n", item->value);
+ } else {
+ g_strlcpy(local_ip, item->value, sizeof(local_ip));
+ local_ip_set = TRUE;
+ }
+ }
+ }
+ }
+
+ item = janus_config_get_item_drilldown(config, "general", "keepalive_interval");
+ if(item && item->value) {
+ keepalive_interval = atoi(item->value);
+ }
+ JANUS_LOG(LOG_VERB, "SIPre keep-alive interval set to %d seconds\n", keepalive_interval);
+
+ item = janus_config_get_item_drilldown(config, "general", "register_ttl");
+ if(item && item->value) {
+ register_ttl = atoi(item->value);
+ }
+ JANUS_LOG(LOG_VERB, "SIPre registration TTL set to %d seconds\n", register_ttl);
+
+ item = janus_config_get_item_drilldown(config, "general", "behind_nat");
+ if(item && item->value) {
+ behind_nat = janus_is_true(item->value);
+ }
+
+ item = janus_config_get_item_drilldown(config, "general", "user_agent");
+ if(item && item->value) {
+ user_agent = g_strdup(item->value);
+ } else {
+ user_agent = g_strdup("Janus WebRTC Gateway SIPre Plugin "JANUS_SIPRE_VERSION_STRING);
+ }
+ JANUS_LOG(LOG_VERB, "SIPre User-Agent set to %s\n", user_agent);
+
+ item = janus_config_get_item_drilldown(config, "general", "events");
+ if(item != NULL && item->value != NULL) {
+ notify_events = janus_is_true(item->value);
+ }
+ if(!notify_events && callback->events_is_enabled()) {
+ JANUS_LOG(LOG_WARN, "Notification of events to handlers disabled for %s\n", JANUS_SIPRE_NAME);
+ }
+
+ janus_config_destroy(config);
+ }
+ config = NULL;
+
+ if(!local_ip_set) {
+ janus_sipre_detect_local_ip(local_ip, sizeof(local_ip));
+ }
+ JANUS_LOG(LOG_VERB, "Local IP set to %s\n", local_ip);
+
+#ifdef HAVE_SRTP_2
+ /* Init randomizer (for randum numbers in SRTP) */
+ RAND_poll();
+#endif
+
+ /* Setup libre */
+ int err = libre_init();
+ if(err) {
+ JANUS_LOG(LOG_ERR, "libre_init() failed: %d (%s)\n", err, strerror(err));
+ return -1;
+ }
+ poll_method_set(poll_method_best());
+ err = sip_alloc(&sipstack, NULL, 32, 32, 32, JANUS_SIPRE_NAME, janus_sipre_cb_exit, NULL);
+ if(err) {
+ JANUS_LOG(LOG_ERR, "Failed to initialize libre SIP stack: %d (%s)\n", err, strerror(err));
+ return -1;
+ }
+ struct sa laddr, laddrs;
+ sa_set_str(&laddr, local_ip, 0);
+ sa_set_str(&laddrs, local_ip, 0);
+ err |= sip_transp_add(sipstack, SIP_TRANSP_UDP, &laddr);
+ err |= sip_transp_add(sipstack, SIP_TRANSP_TCP, &laddr);
+ if(err) {
+ JANUS_LOG(LOG_ERR, "Failed to initialize libre SIP transports: %d (%s)\n", err, strerror(err));
+ return -1;
+ }
+ err = tls_alloc(&tls, TLS_METHOD_SSLV23, NULL, NULL);
+ err |= sip_transp_add(sipstack, SIP_TRANSP_TLS, &laddrs, tls);
+ if(err) {
+ mem_deref(sipstack);
+ mem_deref(tls);
+ JANUS_LOG(LOG_ERR, "Failed to initialize libre SIPS transports: %d (%s)\n", err, strerror(err));
+ return -1;
+ }
+ mem_deref(tls);
+
+ sessions = g_hash_table_new(NULL, NULL);
+ callids = g_hash_table_new(g_str_hash, g_str_equal);
+ identities = g_hash_table_new(g_str_hash, g_str_equal);
+ janus_mutex_init(&sessions_mutex);
+ messages = g_async_queue_new_full((GDestroyNotify) janus_sipre_message_free);
+ /* This is the callback we'll need to invoke to contact the gateway */
+ gateway = callback;
+
+ g_atomic_int_set(&initialized, 1);
+
+ GError *error = NULL;
+ /* Start the sessions watchdog */
+ watchdog = g_thread_try_new("sipre watchdog", &janus_sipre_watchdog, NULL, &error);
+ if(error != NULL) {
+ g_atomic_int_set(&initialized, 0);
+ JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the SIPre watchdog thread...\n", error->code, error->message ? error->message : "??");
+ return -1;
+ }
+ /* Launch the thread that will handle incoming API messages */
+ handler_thread = g_thread_try_new("sipre handler", janus_sipre_handler, NULL, &error);
+ if(error != NULL) {
+ g_atomic_int_set(&initialized, 0);
+ JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the SIPre handler thread...\n", error->code, error->message ? error->message : "??");
+ return -1;
+ }
+ /* Launch the thread that will handle the libre event loop */
+ sipstack_thread = g_thread_try_new("sipre loop", janus_sipre_stack_thread, NULL, &error);
+ if(error != NULL) {
+ g_atomic_int_set(&initialized, 0);
+ JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the SIPre loop thread...\n", error->code, error->message ? error->message : "??");
+ return -1;
+ }
+ JANUS_LOG(LOG_INFO, "%s initialized!\n", JANUS_SIPRE_NAME);
+ return 0;
+}
+
+void janus_sipre_destroy(void) {
+ if(!g_atomic_int_get(&initialized))
+ return;
+ g_atomic_int_set(&stopping, 1);
+
+ g_async_queue_push(messages, &exit_message);
+ if(handler_thread != NULL) {
+ g_thread_join(handler_thread);
+ handler_thread = NULL;
+ }
+ re_cancel();
+ JANUS_LOG(LOG_ERR, "re_cancel() called\n");
+ if(sipstack_thread != NULL) {
+ g_thread_join(sipstack_thread);
+ sipstack_thread = NULL;
+ }
+ if(watchdog != NULL) {
+ g_thread_join(watchdog);
+ watchdog = NULL;
+ }
+ /* FIXME We should destroy the sessions cleanly */
+ janus_mutex_lock(&sessions_mutex);
+ g_hash_table_destroy(sessions);
+ g_hash_table_destroy(callids);
+ g_hash_table_destroy(identities);
+ sessions = NULL;
+ callids = NULL;
+ identities = NULL;
+ janus_mutex_unlock(&sessions_mutex);
+ g_async_queue_unref(messages);
+ messages = NULL;
+ g_atomic_int_set(&initialized, 0);
+ g_atomic_int_set(&stopping, 0);
+
+ /* Deinitialize libre */
+ libre_close();
+ tmr_debug();
+ mem_debug();
+
+ JANUS_LOG(LOG_INFO, "%s destroyed!\n", JANUS_SIPRE_NAME);
+}
+
+int janus_sipre_get_api_compatibility(void) {
+ /* Important! This is what your plugin MUST always return: don't lie here or bad things will happen */
+ return JANUS_PLUGIN_API_VERSION;
+}
+
+int janus_sipre_get_version(void) {
+ return JANUS_SIPRE_VERSION;
+}
+
+const char *janus_sipre_get_version_string(void) {
+ return JANUS_SIPRE_VERSION_STRING;
+}
+
+const char *janus_sipre_get_description(void) {
+ return JANUS_SIPRE_DESCRIPTION;
+}
+
+const char *janus_sipre_get_name(void) {
+ return JANUS_SIPRE_NAME;
+}
+
+const char *janus_sipre_get_author(void) {
+ return JANUS_SIPRE_AUTHOR;
+}
+
+const char *janus_sipre_get_package(void) {
+ return JANUS_SIPRE_PACKAGE;
+}
+
+void janus_sipre_create_session(janus_plugin_session *handle, int *error) {
+ if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized)) {
+ *error = -1;
+ return;
+ }
+ janus_sipre_session *session = g_malloc0(sizeof(janus_sipre_session));
+ session->handle = handle;
+ session->account.identity = NULL;
+ session->account.sips = TRUE;
+ session->account.username = NULL;
+ session->account.display_name = NULL;
+ session->account.user_agent = NULL;
+ session->account.authuser = NULL;
+ session->account.secret = NULL;
+ session->account.secret_type = janus_sipre_secret_type_unknown;
+ session->account.sip_port = 0;
+ session->account.proxy = NULL;
+ session->account.registration_status = janus_sipre_registration_status_unregistered;
+ session->status = janus_sipre_call_status_idle;
+ memset(&session->stack, 0, sizeof(janus_sipre_stack));
+ session->transaction = NULL;
+ session->callee = NULL;
+ session->callid = NULL;
+ session->sdp = NULL;
+ session->media.remote_ip = NULL;
+ session->media.ready = 0;
+ session->media.autoack = TRUE;
+ session->media.require_srtp = FALSE;
+ session->media.has_srtp_local = FALSE;
+ session->media.has_srtp_remote = FALSE;
+ session->media.has_audio = 0;
+ session->media.audio_rtp_fd = -1;
+ session->media.audio_rtcp_fd= -1;
+ session->media.local_audio_rtp_port = 0;
+ session->media.remote_audio_rtp_port = 0;
+ session->media.local_audio_rtcp_port = 0;
+ session->media.remote_audio_rtcp_port = 0;
+ session->media.audio_ssrc = 0;
+ session->media.audio_ssrc_peer = 0;
+ session->media.audio_pt = -1;
+ session->media.audio_pt_name = NULL;
+ session->media.audio_srtp_suite_in = 0;
+ session->media.audio_srtp_suite_out = 0;
+ session->media.audio_send = TRUE;
+ session->media.has_video = 0;
+ session->media.video_rtp_fd = -1;
+ session->media.video_rtcp_fd= -1;
+ session->media.local_video_rtp_port = 0;
+ session->media.remote_video_rtp_port = 0;
+ session->media.local_video_rtcp_port = 0;
+ session->media.remote_video_rtcp_port = 0;
+ session->media.video_ssrc = 0;
+ session->media.video_ssrc_peer = 0;
+ session->media.video_pt = -1;
+ session->media.video_pt_name = NULL;
+ session->media.video_srtp_suite_in = 0;
+ session->media.video_srtp_suite_out = 0;
+ session->media.video_send = TRUE;
+ /* Initialize the RTP context */
+ janus_rtp_switching_context_reset(&session->media.context);
+ session->media.pipefd[0] = -1;
+ session->media.pipefd[1] = -1;
+ session->media.updated = FALSE;
+ janus_mutex_init(&session->rec_mutex);
+ session->destroyed = 0;
+ g_atomic_int_set(&session->hangingup, 0);
+ janus_mutex_init(&session->mutex);
+ handle->plugin_handle = session;
+
+ int err = sipsess_listen(&session->stack.sess_sock, sipstack, 32, janus_sipre_cb_established, session);
+ if(err < 0) {
+ /* TODO Anything we should do? */
+ JANUS_LOG(LOG_ERR, "Error listening: %d (%s)\n", err, strerror(err));
+ }
+
+ janus_mutex_lock(&sessions_mutex);
+ g_hash_table_insert(sessions, handle, session);
+ janus_mutex_unlock(&sessions_mutex);
+
+
+ return;
+}
+
+void janus_sipre_destroy_session(janus_plugin_session *handle, int *error) {
+ if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized)) {
+ *error = -1;
+ return;
+ }
+ janus_sipre_session *session = (janus_sipre_session *)handle->plugin_handle;
+ if(!session) {
+ JANUS_LOG(LOG_ERR, "No SIPre session associated with this handle...\n");
+ *error = -2;
+ return;
+ }
+ janus_mutex_lock(&sessions_mutex);
+ if(!session->destroyed) {
+ g_hash_table_remove(sessions, handle);
+ janus_sipre_hangup_media(handle);
+ session->destroyed = janus_get_monotonic_time();
+ JANUS_LOG(LOG_VERB, "Destroying SIPre session (%s)...\n", session->account.username ? session->account.username : "unregistered user");
+ /* TODO Destroy re-related stuff for this SIP session */
+
+ /* Cleaning up and removing the session is done in a lazy way */
+ old_sessions = g_list_append(old_sessions, session);
+ }
+ janus_mutex_unlock(&sessions_mutex);
+ return;
+}
+
+json_t *janus_sipre_query_session(janus_plugin_session *handle) {
+ if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized)) {
+ return NULL;
+ }
+ janus_sipre_session *session = (janus_sipre_session *)handle->plugin_handle;
+ if(!session) {
+ JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
+ return NULL;
+ }
+ /* Provide some generic info, e.g., if we're in a call and with whom */
+ json_t *info = json_object();
+ json_object_set_new(info, "username", session->account.username ? json_string(session->account.username) : NULL);
+ json_object_set_new(info, "display_name", session->account.display_name ? json_string(session->account.display_name) : NULL);
+ json_object_set_new(info, "user_agent", session->account.user_agent ? json_string(session->account.user_agent) : NULL);
+ json_object_set_new(info, "identity", session->account.identity ? json_string(session->account.identity) : NULL);
+ json_object_set_new(info, "registration_status", json_string(janus_sipre_registration_status_string(session->account.registration_status)));
+ json_object_set_new(info, "call_status", json_string(janus_sipre_call_status_string(session->status)));
+ if(session->callee) {
+ json_object_set_new(info, "callee", json_string(session->callee ? session->callee : "??"));
+ json_object_set_new(info, "auto-ack", json_string(session->media.autoack ? "yes" : "no"));
+ json_object_set_new(info, "srtp-required", json_string(session->media.require_srtp ? "yes" : "no"));
+ json_object_set_new(info, "sdes-local", json_string(session->media.has_srtp_local ? "yes" : "no"));
+ json_object_set_new(info, "sdes-remote", json_string(session->media.has_srtp_remote ? "yes" : "no"));
+ }
+ if(session->arc || session->vrc || session->arc_peer || session->vrc_peer) {
+ json_t *recording = json_object();
+ if(session->arc && session->arc->filename)
+ json_object_set_new(recording, "audio", json_string(session->arc->filename));
+ if(session->vrc && session->vrc->filename)
+ json_object_set_new(recording, "video", json_string(session->vrc->filename));
+ if(session->arc_peer && session->arc_peer->filename)
+ json_object_set_new(recording, "audio-peer", json_string(session->arc_peer->filename));
+ if(session->vrc_peer && session->vrc_peer->filename)
+ json_object_set_new(recording, "video-peer", json_string(session->vrc_peer->filename));
+ json_object_set_new(info, "recording", recording);
+ }
+ json_object_set_new(info, "destroyed", json_integer(session->destroyed));
+ return info;
+}
+
+struct janus_plugin_result *janus_sipre_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep) {
+ if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
+ return janus_plugin_result_new(JANUS_PLUGIN_ERROR, g_atomic_int_get(&stopping) ? "Shutting down" : "Plugin not initialized", NULL);
+ janus_sipre_message *msg = g_malloc0(sizeof(janus_sipre_message));
+ msg->handle = handle;
+ msg->transaction = transaction;
+ msg->message = message;
+ msg->jsep = jsep;
+ g_async_queue_push(messages, msg);
+
+ /* All the requests to this plugin are handled asynchronously */
+ return janus_plugin_result_new(JANUS_PLUGIN_OK_WAIT, NULL, NULL);
+}
+
+void janus_sipre_setup_media(janus_plugin_session *handle) {
+ JANUS_LOG(LOG_INFO, "WebRTC media is now available\n");
+ if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
+ return;
+ janus_sipre_session *session = (janus_sipre_session *)handle->plugin_handle;
+ if(!session) {
+ JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
+ return;
+ }
+ if(session->destroyed)
+ return;
+ g_atomic_int_set(&session->hangingup, 0);
+ /* TODO Only relay RTP/RTCP when we get this event */
+}
+
+void janus_sipre_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+ if(handle == NULL || handle->stopped || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
+ return;
+ if(gateway) {
+ /* Honour the audio/video active flags */
+ janus_sipre_session *session = (janus_sipre_session *)handle->plugin_handle;
+ if(!session || session->destroyed) {
+ JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
+ return;
+ }
+ /* Forward to our SIPre peer */
+ if((video && !session->media.video_send) || (!video && !session->media.audio_send)) {
+ /* Dropping packet, peer doesn't want to receive it */
+ return;
+ }
+ if((video && session->media.video_ssrc == 0) || (!video && session->media.audio_ssrc == 0)) {
+ rtp_header *header = (rtp_header *)buf;
+ if(video) {
+ session->media.video_ssrc = ntohl(header->ssrc);
+ } else {
+ session->media.audio_ssrc = ntohl(header->ssrc);
+ }
+ JANUS_LOG(LOG_VERB, "[SIPre-%s] Got SIPre %s SSRC: %"SCNu32"\n",
+ session->account.username ? session->account.username : "unknown",
+ video ? "video" : "audio",
+ video ? session->media.video_ssrc : session->media.audio_ssrc);
+ }
+ if((video && session->media.has_video && session->media.video_rtp_fd) ||
+ (!video && session->media.has_audio && session->media.audio_rtp_fd)) {
+ /* Save the frame if we're recording */
+ janus_recorder_save_frame(video ? session->vrc : session->arc, buf, len);
+ /* Is SRTP involved? */
+ if(session->media.has_srtp_local) {
+ char sbuf[2048];
+ memcpy(&sbuf, buf, len);
+ int protected = len;
+ int res = srtp_protect(
+ (video ? session->media.video_srtp_out : session->media.audio_srtp_out),
+ &sbuf, &protected);
+ if(res != srtp_err_status_ok) {
+ rtp_header *header = (rtp_header *)&sbuf;
+ guint32 timestamp = ntohl(header->timestamp);
+ guint16 seq = ntohs(header->seq_number);
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] %s SRTP protect error... %s (len=%d-->%d, ts=%"SCNu32", seq=%"SCNu16")...\n",
+ session->account.username ? session->account.username : "unknown",
+ video ? "Video" : "Audio", janus_srtp_error_str(res), len, protected, timestamp, seq);
+ } else {
+ /* Forward the frame to the peer */
+ send((video ? session->media.video_rtp_fd : session->media.audio_rtp_fd), sbuf, protected, 0);
+ }
+ } else {
+ /* Forward the frame to the peer */
+ send((video ? session->media.video_rtp_fd : session->media.audio_rtp_fd), buf, len, 0);
+ }
+ }
+ }
+}
+
+void janus_sipre_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+ if(handle == NULL || handle->stopped || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
+ return;
+ if(gateway) {
+ janus_sipre_session *session = (janus_sipre_session *)handle->plugin_handle;
+ if(!session || session->destroyed) {
+ JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
+ return;
+ }
+ /* Forward to our SIPre peer */
+ if((video && session->media.has_video && session->media.video_rtcp_fd) ||
+ (!video && session->media.has_audio && session->media.audio_rtcp_fd)) {
+ /* Fix SSRCs as the gateway does */
+ JANUS_LOG(LOG_HUGE, "[SIPre-%s] Fixing %s SSRCs (local %u, peer %u)\n",
+ session->account.username ? session->account.username : "unknown",
+ video ? "video" : "audio",
+ (video ? session->media.video_ssrc : session->media.audio_ssrc),
+ (video ? session->media.video_ssrc_peer : session->media.audio_ssrc_peer));
+ janus_rtcp_fix_ssrc(NULL, (char *)buf, len, video,
+ (video ? session->media.video_ssrc : session->media.audio_ssrc),
+ (video ? session->media.video_ssrc_peer : session->media.audio_ssrc_peer));
+ /* Is SRTP involved? */
+ if(session->media.has_srtp_local) {
+ char sbuf[2048];
+ memcpy(&sbuf, buf, len);
+ int protected = len;
+ int res = srtp_protect_rtcp(
+ (video ? session->media.video_srtp_out : session->media.audio_srtp_out),
+ &sbuf, &protected);
+ if(res != srtp_err_status_ok) {
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] %s SRTCP protect error... %s (len=%d-->%d)...\n",
+ session->account.username ? session->account.username : "unknown",
+ video ? "Video" : "Audio",
+ janus_srtp_error_str(res), len, protected);
+ } else {
+ /* Forward the message to the peer */
+ send((video ? session->media.video_rtcp_fd : session->media.audio_rtcp_fd), sbuf, protected, 0);
+ }
+ } else {
+ /* Forward the message to the peer */
+ send((video ? session->media.video_rtcp_fd : session->media.audio_rtcp_fd), buf, len, 0);
+ }
+ }
+ }
+}
+
+void janus_sipre_hangup_media(janus_plugin_session *handle) {
+ JANUS_LOG(LOG_INFO, "No WebRTC media anymore\n");
+ if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
+ return;
+ janus_sipre_session *session = (janus_sipre_session *)handle->plugin_handle;
+ if(!session) {
+ JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
+ return;
+ }
+ if(session->destroyed)
+ return;
+ if(g_atomic_int_add(&session->hangingup, 1))
+ return;
+ if(!(session->status == janus_sipre_call_status_inviting ||
+ session->status == janus_sipre_call_status_invited ||
+ session->status == janus_sipre_call_status_incall))
+ return;
+ /* Get rid of the recorders, if available */
+ janus_mutex_lock(&session->rec_mutex);
+ if(session->arc) {
+ janus_recorder_close(session->arc);
+ JANUS_LOG(LOG_INFO, "Closed user's audio recording %s\n", session->arc->filename ? session->arc->filename : "??");
+ janus_recorder_free(session->arc);
+ }
+ session->arc = NULL;
+ if(session->arc_peer) {
+ janus_recorder_close(session->arc_peer);
+ JANUS_LOG(LOG_INFO, "Closed peer's audio recording %s\n", session->arc_peer->filename ? session->arc_peer->filename : "??");
+ janus_recorder_free(session->arc_peer);
+ }
+ session->arc_peer = NULL;
+ if(session->vrc) {
+ janus_recorder_close(session->vrc);
+ JANUS_LOG(LOG_INFO, "Closed user's video recording %s\n", session->vrc->filename ? session->vrc->filename : "??");
+ janus_recorder_free(session->vrc);
+ }
+ session->vrc = NULL;
+ if(session->vrc_peer) {
+ janus_recorder_close(session->vrc_peer);
+ JANUS_LOG(LOG_INFO, "Closed peer's video recording %s\n", session->vrc_peer->filename ? session->vrc_peer->filename : "??");
+ janus_recorder_free(session->vrc_peer);
+ }
+ session->vrc_peer = NULL;
+ janus_mutex_unlock(&session->rec_mutex);
+ /* FIXME Simulate a "hangup" coming from the browser */
+ janus_sipre_message *msg = g_malloc0(sizeof(janus_sipre_message));
+ msg->handle = handle;
+ msg->message = json_pack("{ss}", "request", "hangup");
+ msg->transaction = NULL;
+ msg->jsep = NULL;
+ g_async_queue_push(messages, msg);
+}
+
+/* Thread to handle incoming messages */
+static void *janus_sipre_handler(void *data) {
+ JANUS_LOG(LOG_VERB, "Joining SIPre handler thread\n");
+ janus_sipre_message *msg = NULL;
+ int error_code = 0;
+ char error_cause[512];
+ json_t *root = NULL;
+ while(g_atomic_int_get(&initialized) && !g_atomic_int_get(&stopping)) {
+ msg = g_async_queue_pop(messages);
+ if(msg == NULL)
+ continue;
+ if(msg == &exit_message)
+ break;
+ if(msg->handle == NULL) {
+ janus_sipre_message_free(msg);
+ continue;
+ }
+ janus_sipre_session *session = NULL;
+ janus_mutex_lock(&sessions_mutex);
+ if(g_hash_table_lookup(sessions, msg->handle) != NULL ) {
+ session = (janus_sipre_session *)msg->handle->plugin_handle;
+ }
+ janus_mutex_unlock(&sessions_mutex);
+ if(!session) {
+ JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
+ janus_sipre_message_free(msg);
+ continue;
+ }
+ if(session->destroyed) {
+ janus_sipre_message_free(msg);
+ continue;
+ }
+ /* Handle request */
+ error_code = 0;
+ root = msg->message;
+ if(msg->message == NULL) {
+ JANUS_LOG(LOG_ERR, "No message??\n");
+ error_code = JANUS_SIPRE_ERROR_NO_MESSAGE;
+ g_snprintf(error_cause, 512, "%s", "No message??");
+ goto error;
+ }
+ if(!json_is_object(root)) {
+ JANUS_LOG(LOG_ERR, "JSON error: not an object\n");
+ error_code = JANUS_SIPRE_ERROR_INVALID_JSON;
+ g_snprintf(error_cause, 512, "JSON error: not an object");
+ goto error;
+ }
+ JANUS_VALIDATE_JSON_OBJECT(root, request_parameters,
+ error_code, error_cause, TRUE,
+ JANUS_SIPRE_ERROR_MISSING_ELEMENT, JANUS_SIPRE_ERROR_INVALID_ELEMENT);
+ if(error_code != 0)
+ goto error;
+ json_t *request = json_object_get(root, "request");
+ const char *request_text = json_string_value(request);
+ json_t *result = NULL;
+
+ if(!strcasecmp(request_text, "register")) {
+ /* Send a REGISTER */
+ if(session->account.registration_status > janus_sipre_registration_status_unregistered) {
+ JANUS_LOG(LOG_ERR, "Already registered (%s)\n", session->account.username);
+ error_code = JANUS_SIPRE_ERROR_ALREADY_REGISTERED;
+ g_snprintf(error_cause, 512, "Already registered (%s)", session->account.username);
+ goto error;
+ }
+
+ /* Cleanup old values */
+ if(session->account.identity != NULL) {
+ g_hash_table_remove(identities, session->account.identity);
+ g_free(session->account.identity);
+ }
+ session->account.identity = NULL;
+ session->account.sips = TRUE;
+ if(session->account.username != NULL)
+ g_free(session->account.username);
+ session->account.username = NULL;
+ if(session->account.display_name != NULL)
+ g_free(session->account.display_name);
+ session->account.display_name = NULL;
+ if(session->account.authuser != NULL)
+ g_free(session->account.authuser);
+ session->account.authuser = NULL;
+ if(session->account.secret != NULL)
+ g_free(session->account.secret);
+ session->account.secret = NULL;
+ session->account.secret_type = janus_sipre_secret_type_unknown;
+ if(session->account.proxy != NULL)
+ g_free(session->account.proxy);
+ session->account.proxy = NULL;
+ if(session->account.user_agent != NULL)
+ g_free(session->account.user_agent);
+ session->account.user_agent = NULL;
+ session->account.registration_status = janus_sipre_registration_status_unregistered;
+
+ gboolean guest = FALSE;
+ JANUS_VALIDATE_JSON_OBJECT(root, register_parameters,
+ error_code, error_cause, TRUE,
+ JANUS_SIPRE_ERROR_MISSING_ELEMENT, JANUS_SIPRE_ERROR_INVALID_ELEMENT);
+ if(error_code != 0)
+ goto error;
+ json_t *type = json_object_get(root, "type");
+ if(type != NULL) {
+ const char *type_text = json_string_value(type);
+ if(!strcmp(type_text, "guest")) {
+ JANUS_LOG(LOG_INFO, "Registering as a guest\n");
+ guest = TRUE;
+ } else {
+ JANUS_LOG(LOG_WARN, "Unknown type '%s', ignoring...\n", type_text);
+ }
+ }
+
+ gboolean send_register = TRUE;
+ json_t *do_register = json_object_get(root, "send_register");
+ if(do_register != NULL) {
+ if(guest) {
+ JANUS_LOG(LOG_ERR, "Conflicting elements: send_register cannot be true if guest is true\n");
+ error_code = JANUS_SIPRE_ERROR_INVALID_ELEMENT;
+ g_snprintf(error_cause, 512, "Conflicting elements: send_register cannot be true if guest is true");
+ goto error;
+ }
+ send_register = json_is_true(do_register);
+ }
+
+ gboolean sips = TRUE;
+ json_t *do_sipres = json_object_get(root, "sips");
+ if(do_sipres != NULL) {
+ sips = json_is_true(do_sipres);
+ }
+
+ /* Parse address */
+ json_t *proxy = json_object_get(root, "proxy");
+ const char *proxy_text = NULL;
+ if(proxy && !json_is_null(proxy)) {
+ /* Has to be validated separately because it could be null */
+ JANUS_VALIDATE_JSON_OBJECT(root, proxy_parameters,
+ error_code, error_cause, TRUE,
+ JANUS_SIPRE_ERROR_MISSING_ELEMENT, JANUS_SIPRE_ERROR_INVALID_ELEMENT);
+ if(error_code != 0)
+ goto error;
+ proxy_text = json_string_value(proxy);
+ if(janus_sipre_parse_uri(proxy_text) < 0) {
+ JANUS_LOG(LOG_ERR, "Invalid proxy address %s\n", proxy_text);
+ error_code = JANUS_SIPRE_ERROR_INVALID_ADDRESS;
+ g_snprintf(error_cause, 512, "Invalid proxy address %s\n", proxy_text);
+ goto error;
+ }
+ }
+
+ /* Parse register TTL */
+ int ttl = register_ttl;
+ json_t *reg_ttl = json_object_get(root, "register_ttl");
+ if(reg_ttl && json_is_integer(reg_ttl))
+ ttl = json_integer_value(reg_ttl);
+ if(ttl <= 0)
+ ttl = JANUS_DEFAULT_REGISTER_TTL;
+
+ /* Parse display name */
+ const char* display_name_text = NULL;
+ json_t *display_name = json_object_get(root, "display_name");
+ if(display_name && json_is_string(display_name))
+ display_name_text = json_string_value(display_name);
+
+ /* Parse user agent */
+ const char* user_agent_text = NULL;
+ json_t *user_agent = json_object_get(root, "user_agent");
+ if(user_agent && json_is_string(user_agent))
+ user_agent_text = json_string_value(user_agent);
+
+ /* Now the user part, if needed */
+ json_t *username = json_object_get(root, "username");
+ if(!guest && !username) {
+ /* The username is mandatory if we're not registering as guests */
+ JANUS_LOG(LOG_ERR, "Missing element (username)\n");
+ error_code = JANUS_SIPRE_ERROR_MISSING_ELEMENT;
+ g_snprintf(error_cause, 512, "Missing element (username)");
+ goto error;
+ }
+ const char *username_text = NULL;
+ char *user_id = NULL, *user_host = NULL;
+ guint16 user_port = 0;
+ if(username) {
+ /* Parse address */
+ username_text = json_string_value(username);
+ if(janus_sipre_parse_uri(username_text) < 0) {
+ JANUS_LOG(LOG_ERR, "Invalid user address %s\n", username_text);
+ error_code = JANUS_SIPRE_ERROR_INVALID_ADDRESS;
+ g_snprintf(error_cause, 512, "Invalid user address %s\n", username_text);
+ goto error;
+ }
+ user_id = janus_sipre_get_uri_username(username_text);
+ user_host = janus_sipre_get_uri_host(username_text);
+ user_port = janus_sipre_get_uri_port(username_text);
+ }
+ if(guest) {
+ /* Not needed, we can stop here: just pick a random username if it wasn't provided and say we're registered */
+ if(!username)
+ g_snprintf(user_id, 255, "janus-sipre-%"SCNu32"", janus_random_uint32());
+ JANUS_LOG(LOG_INFO, "Guest will have username %s\n", user_id);
+ send_register = FALSE;
+ } else {
+ json_t *secret = json_object_get(root, "secret");
+ json_t *ha1_secret = json_object_get(root, "ha1_secret");
+ json_t *authuser = json_object_get(root, "authuser");
+ if(!secret && !ha1_secret) {
+ g_free(user_id);
+ g_free(user_host);
+ JANUS_LOG(LOG_ERR, "Missing element (secret or ha1_secret)\n");
+ error_code = JANUS_SIPRE_ERROR_MISSING_ELEMENT;
+ g_snprintf(error_cause, 512, "Missing element (secret or ha1_secret)");
+ goto error;
+ }
+ if(secret && ha1_secret) {
+ g_free(user_id);
+ g_free(user_host);
+ JANUS_LOG(LOG_ERR, "Conflicting elements specified (secret and ha1_secret)\n");
+ error_code = JANUS_SIPRE_ERROR_INVALID_ELEMENT;
+ g_snprintf(error_cause, 512, "Conflicting elements specified (secret and ha1_secret)");
+ goto error;
+ }
+ const char *secret_text;
+ if(secret) {
+ secret_text = json_string_value(secret);
+ session->account.secret = g_strdup(secret_text);
+ session->account.secret_type = janus_sipre_secret_type_plaintext;
+ } else {
+ secret_text = json_string_value(ha1_secret);
+ session->account.secret = g_strdup(secret_text);
+ session->account.secret_type = janus_sipre_secret_type_hashed;
+ }
+ if(authuser) {
+ const char *authuser_text;
+ authuser_text = json_string_value(authuser);
+ session->account.authuser = g_strdup(authuser_text);
+ } else {
+ session->account.authuser = g_strdup(user_id);
+ }
+ /* Got the values, try registering now */
+ JANUS_LOG(LOG_VERB, "Registering user %s (secret %s) @ %s through %s\n",
+ user_id, secret_text, user_host, proxy_text != NULL ? proxy_text : "(null)");
+ }
+
+ session->account.identity = g_strdup(username_text);
+ g_hash_table_insert(identities, session->account.identity, session);
+ session->account.sips = sips;
+ session->account.username = g_strdup(user_id);
+ if(display_name_text) {
+ session->account.display_name = g_strdup(display_name_text);
+ }
+ if(user_agent_text) {
+ session->account.user_agent = g_strdup(user_agent_text);
+ }
+ if(proxy_text) {
+ session->account.proxy = g_strdup(proxy_text);
+ } else {
+ /* Build one from the user's identity */
+ char uri[256];
+ g_snprintf(uri, sizeof(uri), "sip:%s:%"SCNu16, user_host, (user_port ? user_port : 5060));
+ session->account.proxy = g_strdup(uri);
+ }
+ g_free(user_host);
+ g_free(user_id);
+
+ session->account.registration_status = janus_sipre_registration_status_registering;
+ if(send_register) {
+ char ttl_text[20];
+ g_snprintf(ttl_text, sizeof(ttl_text), "%d", ttl);
+ /* TODO Any way to specify TTL in sipreg_register? */
+ int err = sipreg_register(&session->stack.reg, sipstack,
+ session->account.proxy,
+ session->account.identity, session->account.identity, 3600,
+ session->account.display_name ? session->account.display_name : session->account.username, NULL, 0, 0,
+ janus_sipre_cb_auth, session, FALSE,
+ janus_sipre_cb_register, session, NULL, NULL);
+ if(err < 0) {
+ /* TODO Handle accordingly */
+ JANUS_LOG(LOG_ERR, "Error attempting to REGISTER...\n");
+ }
+ result = json_object();
+ json_object_set_new(result, "event", json_string("registering"));
+ } else {
+ JANUS_LOG(LOG_VERB, "Not sending a SIPre REGISTER: either send_register was set to false or guest mode was enabled\n");
+ session->account.registration_status = janus_sipre_registration_status_disabled;
+ result = json_object();
+ json_object_set_new(result, "event", json_string("registered"));
+ json_object_set_new(result, "username", json_string(session->account.username));
+ json_object_set_new(result, "register_sent", json_false());
+ /* Also notify event handlers */
+ if(notify_events && gateway->events_is_enabled()) {
+ json_t *info = json_object();
+ json_object_set_new(info, "event", json_string("registered"));
+ json_object_set_new(info, "identity", json_string(session->account.identity));
+ json_object_set_new(info, "type", json_string("guest"));
+ gateway->notify_event(&janus_sipre_plugin, session->handle, info);
+ }
+ }
+ } else if(!strcasecmp(request_text, "call")) {
+ /* Call another peer */
+ //~ if(session->stack == NULL) {
+ //~ JANUS_LOG(LOG_ERR, "Wrong state (register first)\n");
+ //~ error_code = JANUS_SIPRE_ERROR_WRONG_STATE;
+ //~ g_snprintf(error_cause, 512, "Wrong state (register first)");
+ //~ goto error;
+ //~ }
+ if(session->status >= janus_sipre_call_status_inviting) {
+ JANUS_LOG(LOG_ERR, "Wrong state (already in a call? status=%s)\n", janus_sipre_call_status_string(session->status));
+ error_code = JANUS_SIPRE_ERROR_WRONG_STATE;
+ g_snprintf(error_cause, 512, "Wrong state (already in a call? status=%s)", janus_sipre_call_status_string(session->status));
+ goto error;
+ }
+ JANUS_VALIDATE_JSON_OBJECT(root, call_parameters,
+ error_code, error_cause, TRUE,
+ JANUS_SIPRE_ERROR_MISSING_ELEMENT, JANUS_SIPRE_ERROR_INVALID_ELEMENT);
+ if(error_code != 0)
+ goto error;
+ json_t *uri = json_object_get(root, "uri");
+ /* Check if we need to ACK manually (e.g., for the Record-Route hack) */
+ json_t *autoack = json_object_get(root, "autoack");
+ gboolean do_autoack = autoack ? json_is_true(autoack) : TRUE;
+ /* Check if the INVITE needs to be enriched with custom headers */
+ char custom_headers[2048];
+ custom_headers[0] = '\0';
+ json_t *headers = json_object_get(root, "headers");
+ if(headers) {
+ if(json_object_size(headers) > 0) {
+ /* Parse custom headers */
+ const char *key = NULL;
+ json_t *value = NULL;
+ void *iter = json_object_iter(headers);
+ while(iter != NULL) {
+ key = json_object_iter_key(iter);
+ value = json_object_get(headers, key);
+ if(value == NULL || !json_is_string(value)) {
+ JANUS_LOG(LOG_WARN, "Skipping header '%s': value is not a string\n", key);
+ iter = json_object_iter_next(headers, iter);
+ continue;
+ }
+ char h[255];
+ g_snprintf(h, 255, "%s: %s\r\n", key, json_string_value(value));
+ JANUS_LOG(LOG_VERB, "Adding custom header, %s", h);
+ g_strlcat(custom_headers, h, 2048);
+ iter = json_object_iter_next(headers, iter);
+ }
+ }
+ }
+ /* SDES-SRTP is disabled by default, let's see if we need to enable it */
+ gboolean offer_srtp = FALSE, require_srtp = FALSE;
+ json_t *srtp = json_object_get(root, "srtp");
+ if(srtp) {
+ const char *srtp_text = json_string_value(srtp);
+ if(!strcasecmp(srtp_text, "sdes_optional")) {
+ /* Negotiate SDES, but make it optional */
+ offer_srtp = TRUE;
+ } else if(!strcasecmp(srtp_text, "sdes_mandatory")) {
+ /* Negotiate SDES, and require it */
+ offer_srtp = TRUE;
+ require_srtp = TRUE;
+ } else {
+ JANUS_LOG(LOG_ERR, "Invalid element (srtp can only be sdes_optional or sdes_mandatory)\n");
+ error_code = JANUS_SIPRE_ERROR_INVALID_ELEMENT;
+ g_snprintf(error_cause, 512, "Invalid element (srtp can only be sdes_optional or sdes_mandatory)");
+ goto error;
+ }
+ }
+ /* Parse address */
+ const char *uri_text = json_string_value(uri);
+ if(janus_sipre_parse_uri(uri_text) < 0) {
+ JANUS_LOG(LOG_ERR, "Invalid user address %s\n", uri_text);
+ error_code = JANUS_SIPRE_ERROR_INVALID_ADDRESS;
+ g_snprintf(error_cause, 512, "Invalid user address %s\n", uri_text);
+ goto error;
+ }
+ /* Any SDP to handle? if not, something's wrong */
+ const char *msg_sdp_type = json_string_value(json_object_get(msg->jsep, "type"));
+ const char *msg_sdp = json_string_value(json_object_get(msg->jsep, "sdp"));
+ if(!msg_sdp) {
+ JANUS_LOG(LOG_ERR, "Missing SDP\n");
+ error_code = JANUS_SIPRE_ERROR_MISSING_SDP;
+ g_snprintf(error_cause, 512, "Missing SDP");
+ goto error;
+ }
+ if(strstr(msg_sdp, "m=application")) {
+ JANUS_LOG(LOG_ERR, "The SIPre plugin does not support DataChannels\n");
+ error_code = JANUS_SIPRE_ERROR_MISSING_SDP;
+ g_snprintf(error_cause, 512, "The SIPre plugin does not support DataChannels");
+ goto error;
+ }
+ JANUS_LOG(LOG_VERB, "%s is calling %s\n", session->account.username, uri_text);
+ JANUS_LOG(LOG_VERB, "This is involving a negotiation (%s) as well:\n%s\n", msg_sdp_type, msg_sdp);
+ /* Clean up SRTP stuff from before first, in case it's still needed */
+ janus_sipre_srtp_cleanup(session);
+ session->media.require_srtp = require_srtp;
+ session->media.has_srtp_local = offer_srtp;
+ if(offer_srtp) {
+ JANUS_LOG(LOG_VERB, "Going to negotiate SDES-SRTP (%s)...\n", require_srtp ? "mandatory" : "optional");
+ }
+ /* Parse the SDP we got, manipulate some things, and generate a new one */
+ char sdperror[100];
+ janus_sdp *parsed_sdp = janus_sdp_parse(msg_sdp, sdperror, sizeof(sdperror));
+ if(!parsed_sdp) {
+ JANUS_LOG(LOG_ERR, "Error parsing SDP: %s\n", sdperror);
+ error_code = JANUS_SIPRE_ERROR_MISSING_SDP;
+ g_snprintf(error_cause, 512, "Error parsing SDP: %s", sdperror);
+ goto error;
+ }
+ /* Allocate RTP ports and merge them with the anonymized SDP */
+ if(strstr(msg_sdp, "m=audio") && !strstr(msg_sdp, "m=audio 0")) {
+ JANUS_LOG(LOG_VERB, "Going to negotiate audio...\n");
+ session->media.has_audio = 1; /* FIXME Maybe we need a better way to signal this */
+ }
+ if(strstr(msg_sdp, "m=video") && !strstr(msg_sdp, "m=video 0")) {
+ JANUS_LOG(LOG_VERB, "Going to negotiate video...\n");
+ session->media.has_video = 1; /* FIXME Maybe we need a better way to signal this */
+ }
+ if(janus_sipre_allocate_local_ports(session) < 0) {
+ JANUS_LOG(LOG_ERR, "Could not allocate RTP/RTCP ports\n");
+ janus_sdp_free(parsed_sdp);
+ error_code = JANUS_SIPRE_ERROR_IO_ERROR;
+ g_snprintf(error_cause, 512, "Could not allocate RTP/RTCP ports");
+ goto error;
+ }
+ char *sdp = janus_sipre_sdp_manipulate(session, parsed_sdp, FALSE);
+ if(sdp == NULL) {
+ JANUS_LOG(LOG_ERR, "Could not allocate RTP/RTCP ports\n");
+ janus_sdp_free(parsed_sdp);
+ error_code = JANUS_SIPRE_ERROR_IO_ERROR;
+ g_snprintf(error_cause, 512, "Could not allocate RTP/RTCP ports");
+ goto error;
+ }
+ /* Take note of the SDP (may be useful for UPDATEs or re-INVITEs) */
+ janus_sdp_free(session->sdp);
+ session->sdp = parsed_sdp;
+ JANUS_LOG(LOG_VERB, "Prepared SDP for INVITE:\n%s", sdp);
+ /* Prepare the From header */
+ char from_hdr[1024];
+ if(session->account.display_name) {
+ g_snprintf(from_hdr, sizeof(from_hdr), "\"%s\" <%s>", session->account.display_name, session->account.identity);
+ } else {
+ g_snprintf(from_hdr, sizeof(from_hdr), "%s", session->account.identity);
+ }
+ /* Prepare the stack */
+ /* TODO */
+ g_atomic_int_set(&session->hangingup, 0);
+ session->status = janus_sipre_call_status_inviting;
+ /* Create a random call-id */
+ char callid[24];
+ janus_sipre_random_string(24, (char *)&callid);
+ /* Also notify event handlers */
+ if(notify_events && gateway->events_is_enabled()) {
+ json_t *info = json_object();
+ json_object_set_new(info, "event", json_string("calling"));
+ json_object_set_new(info, "callee", json_string(uri_text));
+ json_object_set_new(info, "call-id", json_string(callid));
+ json_object_set_new(info, "sdp", json_string(sdp));
+ gateway->notify_event(&janus_sipre_plugin, session->handle, info);
+ }
+ /* Send INVITE */
+ session->callee = g_strdup(uri_text);
+ session->callid = g_strdup(callid);
+ g_hash_table_insert(callids, session->callid, session);
+ session->media.autoack = do_autoack;
+ /* TODO Use re to send INVITE */
+ g_free(sdp);
+ if(session->transaction)
+ g_free(session->transaction);
+ session->transaction = msg->transaction ? g_strdup(msg->transaction) : NULL;
+ /* Send an ack back */
+ result = json_object();
+ json_object_set_new(result, "event", json_string("calling"));
+ } else if(!strcasecmp(request_text, "accept")) {
+ if(session->status != janus_sipre_call_status_invited) {
+ JANUS_LOG(LOG_ERR, "Wrong state (not invited? status=%s)\n", janus_sipre_call_status_string(session->status));
+ error_code = JANUS_SIPRE_ERROR_WRONG_STATE;
+ g_snprintf(error_cause, 512, "Wrong state (not invited? status=%s)", janus_sipre_call_status_string(session->status));
+ goto error;
+ }
+ if(session->callee == NULL) {
+ JANUS_LOG(LOG_ERR, "Wrong state (no caller?)\n");
+ error_code = JANUS_SIPRE_ERROR_WRONG_STATE;
+ g_snprintf(error_cause, 512, "Wrong state (no caller?)");
+ goto error;
+ }
+ JANUS_VALIDATE_JSON_OBJECT(root, accept_parameters,
+ error_code, error_cause, TRUE,
+ JANUS_SIPRE_ERROR_MISSING_ELEMENT, JANUS_SIPRE_ERROR_INVALID_ELEMENT);
+ if(error_code != 0)
+ goto error;
+ json_t *srtp = json_object_get(root, "srtp");
+ gboolean answer_srtp = FALSE;
+ if(srtp) {
+ const char *srtp_text = json_string_value(srtp);
+ if(!strcasecmp(srtp_text, "sdes_optional")) {
+ /* Negotiate SDES, but make it optional */
+ answer_srtp = TRUE;
+ } else if(!strcasecmp(srtp_text, "sdes_mandatory")) {
+ /* Negotiate SDES, and require it */
+ answer_srtp = TRUE;
+ session->media.require_srtp = TRUE;
+ } else {
+ JANUS_LOG(LOG_ERR, "Invalid element (srtp can only be sdes_optional or sdes_mandatory)\n");
+ error_code = JANUS_SIPRE_ERROR_INVALID_ELEMENT;
+ g_snprintf(error_cause, 512, "Invalid element (srtp can only be sdes_optional or sdes_mandatory)");
+ goto error;
+ }
+ }
+ if(session->media.require_srtp && !session->media.has_srtp_remote) {
+ JANUS_LOG(LOG_ERR, "Can't accept the call: SDES-SRTP required, but caller didn't offer it\n");
+ error_code = JANUS_SIPRE_ERROR_TOO_STRICT;
+ g_snprintf(error_cause, 512, "Can't accept the call: SDES-SRTP required, but caller didn't offer it");
+ goto error;
+ }
+ answer_srtp = answer_srtp || session->media.has_srtp_remote;
+ /* Any SDP to handle? if not, something's wrong */
+ const char *msg_sdp_type = json_string_value(json_object_get(msg->jsep, "type"));
+ const char *msg_sdp = json_string_value(json_object_get(msg->jsep, "sdp"));
+ if(!msg_sdp) {
+ JANUS_LOG(LOG_ERR, "Missing SDP\n");
+ error_code = JANUS_SIPRE_ERROR_MISSING_SDP;
+ g_snprintf(error_cause, 512, "Missing SDP");
+ goto error;
+ }
+ /* Accept a call from another peer */
+ JANUS_LOG(LOG_VERB, "We're accepting the call from %s\n", session->callee);
+ JANUS_LOG(LOG_VERB, "This is involving a negotiation (%s) as well:\n%s\n", msg_sdp_type, msg_sdp);
+ session->media.has_srtp_local = answer_srtp;
+ if(answer_srtp) {
+ JANUS_LOG(LOG_VERB, "Going to negotiate SDES-SRTP (%s)...\n", session->media.require_srtp ? "mandatory" : "optional");
+ }
+ /* Parse the SDP we got, manipulate some things, and generate a new one */
+ char sdperror[100];
+ janus_sdp *parsed_sdp = janus_sdp_parse(msg_sdp, sdperror, sizeof(sdperror));
+ if(!parsed_sdp) {
+ JANUS_LOG(LOG_ERR, "Error parsing SDP: %s\n", sdperror);
+ error_code = JANUS_SIPRE_ERROR_MISSING_SDP;
+ g_snprintf(error_cause, 512, "Error parsing SDP: %s", sdperror);
+ goto error;
+ }
+ /* Allocate RTP ports and merge them with the anonymized SDP */
+ if(strstr(msg_sdp, "m=audio") && !strstr(msg_sdp, "m=audio 0")) {
+ JANUS_LOG(LOG_VERB, "Going to negotiate audio...\n");
+ session->media.has_audio = 1; /* FIXME Maybe we need a better way to signal this */
+ }
+ if(strstr(msg_sdp, "m=video") && !strstr(msg_sdp, "m=video 0")) {
+ JANUS_LOG(LOG_VERB, "Going to negotiate video...\n");
+ session->media.has_video = 1; /* FIXME Maybe we need a better way to signal this */
+ }
+ if(janus_sipre_allocate_local_ports(session) < 0) {
+ JANUS_LOG(LOG_ERR, "Could not allocate RTP/RTCP ports\n");
+ janus_sdp_free(parsed_sdp);
+ error_code = JANUS_SIPRE_ERROR_IO_ERROR;
+ g_snprintf(error_cause, 512, "Could not allocate RTP/RTCP ports");
+ goto error;
+ }
+ char *sdp = janus_sipre_sdp_manipulate(session, parsed_sdp, TRUE);
+ if(sdp == NULL) {
+ JANUS_LOG(LOG_ERR, "Could not allocate RTP/RTCP ports\n");
+ janus_sdp_free(parsed_sdp);
+ error_code = JANUS_SIPRE_ERROR_IO_ERROR;
+ g_snprintf(error_cause, 512, "Could not allocate RTP/RTCP ports");
+ goto error;
+ }
+ if(session->media.audio_pt > -1) {
+ session->media.audio_pt_name = janus_get_codec_from_pt(sdp, session->media.audio_pt);
+ JANUS_LOG(LOG_VERB, "Detected audio codec: %d (%s)\n", session->media.audio_pt, session->media.audio_pt_name);
+ }
+ if(session->media.video_pt > -1) {
+ session->media.video_pt_name = janus_get_codec_from_pt(sdp, session->media.video_pt);
+ JANUS_LOG(LOG_VERB, "Detected video codec: %d (%s)\n", session->media.video_pt, session->media.video_pt_name);
+ }
+ /* Take note of the SDP (may be useful for UPDATEs or re-INVITEs) */
+ janus_sdp_free(session->sdp);
+ session->sdp = parsed_sdp;
+ JANUS_LOG(LOG_VERB, "Prepared SDP for 200 OK:\n%s", sdp);
+ /* Also notify event handlers */
+ if(notify_events && gateway->events_is_enabled()) {
+ json_t *info = json_object();
+ json_object_set_new(info, "event", json_string("accepted"));
+ if(session->callid)
+ json_object_set_new(info, "call-id", json_string(session->callid));
+ gateway->notify_event(&janus_sipre_plugin, session->handle, info);
+ }
+ /* Send 200 OK */
+ g_atomic_int_set(&session->hangingup, 0);
+ session->status = janus_sipre_call_status_incall;
+ /* TODO Use re to send 200 OK */
+ g_free(sdp);
+ /* Send an ack back */
+ result = json_object();
+ json_object_set_new(result, "event", json_string("accepted"));
+ /* Start the media */
+ session->media.ready = 1; /* FIXME Maybe we need a better way to signal this */
+ GError *error = NULL;
+ char tname[16];
+ g_snprintf(tname, sizeof(tname), "siprtp %s", session->account.username);
+ g_thread_try_new(tname, janus_sipre_relay_thread, session, &error);
+ if(error != NULL) {
+ JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the RTP/RTCP thread...\n", error->code, error->message ? error->message : "??");
+ }
+ } else if(!strcasecmp(request_text, "decline")) {
+ /* Reject an incoming call */
+ if(session->status != janus_sipre_call_status_invited) {
+ JANUS_LOG(LOG_ERR, "Wrong state (not invited? status=%s)\n", janus_sipre_call_status_string(session->status));
+ /* Ignore */
+ janus_sipre_message_free(msg);
+ continue;
+ //~ g_snprintf(error_cause, 512, "Wrong state (not in a call?)");
+ //~ goto error;
+ }
+ if(session->callee == NULL) {
+ JANUS_LOG(LOG_ERR, "Wrong state (no callee?)\n");
+ error_code = JANUS_SIPRE_ERROR_WRONG_STATE;
+ g_snprintf(error_cause, 512, "Wrong state (no callee?)");
+ goto error;
+ }
+ session->status = janus_sipre_call_status_closing;
+ /* Prepare response */
+ int response_code = 486;
+ json_t *code_json = json_object_get(root, "code");
+ if(code_json && json_is_integer(code_json))
+ response_code = json_integer_value(code_json);
+ if(response_code <= 399) {
+ JANUS_LOG(LOG_WARN, "Invalid SIPre response code specified, using 486 to decline call\n");
+ response_code = 486;
+ }
+ /* TODO Use re to send error */
+ /* Also notify event handlers */
+ if(notify_events && gateway->events_is_enabled()) {
+ json_t *info = json_object();
+ json_object_set_new(info, "event", json_string("declined"));
+ json_object_set_new(info, "callee", json_string(session->callee));
+ if(session->callid)
+ json_object_set_new(info, "call-id", json_string(session->callid));
+ json_object_set_new(info, "code", json_integer(response_code));
+ gateway->notify_event(&janus_sipre_plugin, session->handle, info);
+ }
+ g_free(session->callee);
+ session->callee = NULL;
+ /* Notify the operation */
+ result = json_object();
+ json_object_set_new(result, "event", json_string("declining"));
+ json_object_set_new(result, "code", json_integer(response_code));
+ } else if(!strcasecmp(request_text, "hangup")) {
+ /* Hangup an ongoing call */
+ if(!(session->status == janus_sipre_call_status_inviting || session->status == janus_sipre_call_status_incall)) {
+ JANUS_LOG(LOG_ERR, "Wrong state (not in a call? status=%s)\n", janus_sipre_call_status_string(session->status));
+ /* Ignore */
+ janus_sipre_message_free(msg);
+ continue;
+ //~ g_snprintf(error_cause, 512, "Wrong state (not in a call?)");
+ //~ goto error;
+ }
+ if(session->callee == NULL) {
+ JANUS_LOG(LOG_ERR, "Wrong state (no callee?)\n");
+ error_code = JANUS_SIPRE_ERROR_WRONG_STATE;
+ g_snprintf(error_cause, 512, "Wrong state (no callee?)");
+ goto error;
+ }
+ session->status = janus_sipre_call_status_closing;
+ /* TODO Use re to send BYE */
+ g_free(session->callee);
+ session->callee = NULL;
+ /* Notify the operation */
+ result = json_object();
+ json_object_set_new(result, "event", json_string("hangingup"));
+ } else if(!strcasecmp(request_text, "recording")) {
+ /* Start or stop recording */
+ if(!(session->status == janus_sipre_call_status_inviting || session->status == janus_sipre_call_status_incall)) {
+ JANUS_LOG(LOG_ERR, "Wrong state (not in a call? status=%s)\n", janus_sipre_call_status_string(session->status));
+ g_snprintf(error_cause, 512, "Wrong state (not in a call?)");
+ goto error;
+ }
+ if(session->callee == NULL) {
+ JANUS_LOG(LOG_ERR, "Wrong state (no callee?)\n");
+ error_code = JANUS_SIPRE_ERROR_WRONG_STATE;
+ g_snprintf(error_cause, 512, "Wrong state (no callee?)");
+ goto error;
+ }
+ JANUS_VALIDATE_JSON_OBJECT(root, recording_parameters,
+ error_code, error_cause, TRUE,
+ JANUS_SIPRE_ERROR_MISSING_ELEMENT, JANUS_SIPRE_ERROR_INVALID_ELEMENT);
+ if(error_code != 0)
+ goto error;
+ json_t *action = json_object_get(root, "action");
+ const char *action_text = json_string_value(action);
+ if(strcasecmp(action_text, "start") && strcasecmp(action_text, "stop")) {
+ JANUS_LOG(LOG_ERR, "Invalid action (should be start|stop)\n");
+ error_code = JANUS_SIPRE_ERROR_INVALID_ELEMENT;
+ g_snprintf(error_cause, 512, "Invalid action (should be start|stop)");
+ goto error;
+ }
+ gboolean record_audio = FALSE, record_video = FALSE, /* No media is recorded by default */
+ record_peer_audio = FALSE, record_peer_video = FALSE;
+ json_t *audio = json_object_get(root, "audio");
+ record_audio = audio ? json_is_true(audio) : FALSE;
+ json_t *video = json_object_get(root, "video");
+ record_video = video ? json_is_true(video) : FALSE;
+ json_t *peer_audio = json_object_get(root, "peer_audio");
+ record_peer_audio = peer_audio ? json_is_true(peer_audio) : FALSE;
+ json_t *peer_video = json_object_get(root, "peer_video");
+ record_peer_video = peer_video ? json_is_true(peer_video) : FALSE;
+ if(!record_audio && !record_video && !record_peer_audio && !record_peer_video) {
+ JANUS_LOG(LOG_ERR, "Invalid request (at least one of audio, video, peer_audio and peer_video should be true)\n");
+ error_code = JANUS_SIPRE_ERROR_RECORDING_ERROR;
+ g_snprintf(error_cause, 512, "Invalid request (at least one of audio, video, peer_audio and peer_video should be true)");
+ goto error;
+ }
+ json_t *recfile = json_object_get(root, "filename");
+ const char *recording_base = json_string_value(recfile);
+ janus_mutex_lock(&session->rec_mutex);
+ if(!strcasecmp(action_text, "start")) {
+ /* Start recording something */
+ char filename[255];
+ gint64 now = janus_get_real_time();
+ if(record_peer_audio || record_peer_video) {
+ JANUS_LOG(LOG_INFO, "Starting recording of peer's %s (user %s, call %s)\n",
+ (record_peer_audio && record_peer_video ? "audio and video" : (record_peer_audio ? "audio" : "video")),
+ session->account.username, session->transaction);
+ /* Start recording this peer's audio and/or video */
+ if(record_peer_audio) {
+ memset(filename, 0, 255);
+ if(recording_base) {
+ /* Use the filename and path we have been provided */
+ g_snprintf(filename, 255, "%s-peer-audio", recording_base);
+ /* FIXME This only works if offer/answer happened */
+ session->arc_peer = janus_recorder_create(NULL, session->media.audio_pt_name, filename);
+ if(session->arc_peer == NULL) {
+ /* FIXME We should notify the fact the recorder could not be created */
+ JANUS_LOG(LOG_ERR, "Couldn't open an audio recording file for this peer!\n");
+ }
+ } else {
+ /* Build a filename */
+ g_snprintf(filename, 255, "sip-%s-%s-%"SCNi64"-peer-audio",
+ session->account.username ? session->account.username : "unknown",
+ session->transaction ? session->transaction : "unknown",
+ now);
+ /* FIXME This only works if offer/answer happened */
+ session->arc_peer = janus_recorder_create(NULL, session->media.audio_pt_name, filename);
+ if(session->arc_peer == NULL) {
+ /* FIXME We should notify the fact the recorder could not be created */
+ JANUS_LOG(LOG_ERR, "Couldn't open an audio recording file for this peer!\n");
+ }
+ }
+ }
+ if(record_peer_video) {
+ memset(filename, 0, 255);
+ if(recording_base) {
+ /* Use the filename and path we have been provided */
+ g_snprintf(filename, 255, "%s-peer-video", recording_base);
+ /* FIXME This only works if offer/answer happened */
+ session->vrc_peer = janus_recorder_create(NULL, session->media.video_pt_name, filename);
+ if(session->vrc_peer == NULL) {
+ /* FIXME We should notify the fact the recorder could not be created */
+ JANUS_LOG(LOG_ERR, "Couldn't open an video recording file for this peer!\n");
+ }
+ } else {
+ /* Build a filename */
+ g_snprintf(filename, 255, "sip-%s-%s-%"SCNi64"-peer-video",
+ session->account.username ? session->account.username : "unknown",
+ session->transaction ? session->transaction : "unknown",
+ now);
+ /* FIXME This only works if offer/answer happened */
+ session->vrc_peer = janus_recorder_create(NULL, session->media.video_pt_name, filename);
+ if(session->vrc_peer == NULL) {
+ /* FIXME We should notify the fact the recorder could not be created */
+ JANUS_LOG(LOG_ERR, "Couldn't open an video recording file for this peer!\n");
+ }
+ }
+ /* TODO We should send a FIR/PLI to this peer... */
+ }
+ }
+ if(record_audio || record_video) {
+ /* Start recording the user's audio and/or video */
+ JANUS_LOG(LOG_INFO, "Starting recording of user's %s (user %s, call %s)\n",
+ (record_audio && record_video ? "audio and video" : (record_audio ? "audio" : "video")),
+ session->account.username, session->transaction);
+ if(record_audio) {
+ memset(filename, 0, 255);
+ if(recording_base) {
+ /* Use the filename and path we have been provided */
+ g_snprintf(filename, 255, "%s-user-audio", recording_base);
+ /* FIXME This only works if offer/answer happened */
+ session->arc = janus_recorder_create(NULL, session->media.audio_pt_name, filename);
+ if(session->arc == NULL) {
+ /* FIXME We should notify the fact the recorder could not be created */
+ JANUS_LOG(LOG_ERR, "Couldn't open an audio recording file for this peer!\n");
+ }
+ } else {
+ /* Build a filename */
+ g_snprintf(filename, 255, "sip-%s-%s-%"SCNi64"-own-audio",
+ session->account.username ? session->account.username : "unknown",
+ session->transaction ? session->transaction : "unknown",
+ now);
+ /* FIXME This only works if offer/answer happened */
+ session->arc = janus_recorder_create(NULL, session->media.audio_pt_name, filename);
+ if(session->arc == NULL) {
+ /* FIXME We should notify the fact the recorder could not be created */
+ JANUS_LOG(LOG_ERR, "Couldn't open an audio recording file for this peer!\n");
+ }
+ }
+ }
+ if(record_video) {
+ memset(filename, 0, 255);
+ if(recording_base) {
+ /* Use the filename and path we have been provided */
+ g_snprintf(filename, 255, "%s-user-video", recording_base);
+ /* FIXME This only works if offer/answer happened */
+ session->vrc = janus_recorder_create(NULL, session->media.video_pt_name, filename);
+ if(session->vrc == NULL) {
+ /* FIXME We should notify the fact the recorder could not be created */
+ JANUS_LOG(LOG_ERR, "Couldn't open an video recording file for this user!\n");
+ }
+ } else {
+ /* Build a filename */
+ g_snprintf(filename, 255, "sip-%s-%s-%"SCNi64"-own-video",
+ session->account.username ? session->account.username : "unknown",
+ session->transaction ? session->transaction : "unknown",
+ now);
+ /* FIXME This only works if offer/answer happened */
+ session->vrc = janus_recorder_create(NULL, session->media.video_pt_name, filename);
+ if(session->vrc == NULL) {
+ /* FIXME We should notify the fact the recorder could not be created */
+ JANUS_LOG(LOG_ERR, "Couldn't open an video recording file for this user!\n");
+ }
+ }
+ /* Send a PLI */
+ JANUS_LOG(LOG_VERB, "Recording video, sending a PLI to kickstart it\n");
+ char buf[12];
+ memset(buf, 0, 12);
+ janus_rtcp_pli((char *)&buf, 12);
+ gateway->relay_rtcp(session->handle, 1, buf, 12);
+ }
+ }
+ } else {
+ /* Stop recording something: notice that this never returns an error, even when we were not recording anything */
+ if(record_audio) {
+ if(session->arc) {
+ janus_recorder_close(session->arc);
+ JANUS_LOG(LOG_INFO, "Closed user's audio recording %s\n", session->arc->filename ? session->arc->filename : "??");
+ janus_recorder_free(session->arc);
+ }
+ session->arc = NULL;
+ }
+ if(record_video) {
+ if(session->vrc) {
+ janus_recorder_close(session->vrc);
+ JANUS_LOG(LOG_INFO, "Closed user's video recording %s\n", session->vrc->filename ? session->vrc->filename : "??");
+ janus_recorder_free(session->vrc);
+ }
+ session->vrc = NULL;
+ }
+ if(record_peer_audio) {
+ if(session->arc_peer) {
+ janus_recorder_close(session->arc_peer);
+ JANUS_LOG(LOG_INFO, "Closed peer's audio recording %s\n", session->arc_peer->filename ? session->arc_peer->filename : "??");
+ janus_recorder_free(session->arc_peer);
+ }
+ session->arc_peer = NULL;
+ }
+ if(record_peer_video) {
+ if(session->vrc_peer) {
+ janus_recorder_close(session->vrc_peer);
+ JANUS_LOG(LOG_INFO, "Closed peer's video recording %s\n", session->vrc_peer->filename ? session->vrc_peer->filename : "??");
+ janus_recorder_free(session->vrc_peer);
+ }
+ session->vrc_peer = NULL;
+ }
+ }
+ janus_mutex_unlock(&session->rec_mutex);
+ /* Notify the result */
+ result = json_object();
+ json_object_set_new(result, "event", json_string("recordingupdated"));
+ } else if(!strcasecmp(request_text, "dtmf_info")) {
+ /* Send DMTF tones using SIPre INFO
+ * (https://tools.ietf.org/html/draft-kaplan-dispatch-info-dtmf-package-00)
+ */
+ if(!(session->status == janus_sipre_call_status_inviting || session->status == janus_sipre_call_status_incall)) {
+ JANUS_LOG(LOG_ERR, "Wrong state (not in a call? status=%s)\n", janus_sipre_call_status_string(session->status));
+ g_snprintf(error_cause, 512, "Wrong state (not in a call?)");
+ goto error;
+ }
+ if(session->callee == NULL) {
+ JANUS_LOG(LOG_ERR, "Wrong state (no callee?)\n");
+ error_code = JANUS_SIPRE_ERROR_WRONG_STATE;
+ g_snprintf(error_cause, 512, "Wrong state (no callee?)");
+ goto error;
+ }
+ JANUS_VALIDATE_JSON_OBJECT(root, dtmf_info_parameters,
+ error_code, error_cause, TRUE,
+ JANUS_SIPRE_ERROR_MISSING_ELEMENT, JANUS_SIPRE_ERROR_INVALID_ELEMENT);
+ if(error_code != 0)
+ goto error;
+ json_t *digit = json_object_get(root, "digit");
+ const char *digit_text = json_string_value(digit);
+ if(strlen(digit_text) != 1) {
+ JANUS_LOG(LOG_ERR, "Invalid element (digit should be one character))\n");
+ error_code = JANUS_SIPRE_ERROR_INVALID_ELEMENT;
+ g_snprintf(error_cause, 512, "Invalid element (digit should be one character)");
+ goto error;
+ }
+ int duration_ms = 0;
+ json_t *duration = json_object_get(root, "duration");
+ duration_ms = duration ? json_integer_value(duration) : 0;
+ if(duration_ms <= 0 || duration_ms > 5000) {
+ duration_ms = 160; /* default value */
+ }
+
+ char payload[64];
+ g_snprintf(payload, sizeof(payload), "Signal=%s\r\nDuration=%d", digit_text, duration_ms);
+ /* TODO Send "application/dtmf-relay" SIP INFO */
+ } else {
+ JANUS_LOG(LOG_ERR, "Unknown request (%s)\n", request_text);
+ error_code = JANUS_SIPRE_ERROR_INVALID_REQUEST;
+ g_snprintf(error_cause, 512, "Unknown request (%s)", request_text);
+ goto error;
+ }
+
+ /* Prepare JSON event */
+ json_t *event = json_object();
+ json_object_set_new(event, "sip", json_string("event"));
+ if(result != NULL)
+ json_object_set_new(event, "result", result);
+ int ret = gateway->push_event(msg->handle, &janus_sipre_plugin, msg->transaction, event, NULL);
+ JANUS_LOG(LOG_VERB, " >> Pushing event: %d (%s)\n", ret, janus_get_api_error(ret));
+ json_decref(event);
+ janus_sipre_message_free(msg);
+ continue;
+
+error:
+ {
+ /* Prepare JSON error event */
+ json_t *event = json_object();
+ json_object_set_new(event, "sip", json_string("event"));
+ json_object_set_new(event, "error_code", json_integer(error_code));
+ json_object_set_new(event, "error", json_string(error_cause));
+ int ret = gateway->push_event(msg->handle, &janus_sipre_plugin, msg->transaction, event, NULL);
+ JANUS_LOG(LOG_VERB, " >> Pushing event: %d (%s)\n", ret, janus_get_api_error(ret));
+ json_decref(event);
+ janus_sipre_message_free(msg);
+ }
+ }
+ JANUS_LOG(LOG_VERB, "Leaving SIPre handler thread\n");
+ return NULL;
+}
+
+
+/* Process an incoming SDP */
+void janus_sipre_sdp_process(janus_sipre_session *session, janus_sdp *sdp, gboolean answer, gboolean update, gboolean *changed) {
+ if(!session || !sdp)
+ return;
+ /* c= */
+ if(sdp->c_addr) {
+ if(update && strcmp(sdp->c_addr, session->media.remote_ip)) {
+ /* This is an update and an address changed */
+ if(changed)
+ *changed = TRUE;
+ }
+ g_free(session->media.remote_ip);
+ session->media.remote_ip = g_strdup(sdp->c_addr);
+ }
+ GList *temp = sdp->m_lines;
+ while(temp) {
+ janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
+ session->media.require_srtp = session->media.require_srtp || (m->proto && !strcasecmp(m->proto, "RTP/SAVP"));
+ if(m->type == JANUS_SDP_AUDIO) {
+ if(m->port) {
+ if(m->port != session->media.remote_audio_rtp_port) {
+ /* This is an update and an address changed */
+ if(changed)
+ *changed = TRUE;
+ }
+ session->media.has_audio = 1;
+ session->media.remote_audio_rtp_port = m->port;
+ session->media.remote_audio_rtcp_port = m->port+1; /* FIXME We're assuming RTCP is on the next port */
+ if(m->direction == JANUS_SDP_SENDONLY || m->direction == JANUS_SDP_INACTIVE)
+ session->media.audio_send = FALSE;
+ else
+ session->media.audio_send = TRUE;
+ } else {
+ session->media.audio_send = FALSE;
+ }
+ } else if(m->type == JANUS_SDP_VIDEO) {
+ if(m->port) {
+ if(m->port != session->media.remote_video_rtp_port) {
+ /* This is an update and an address changed */
+ if(changed)
+ *changed = TRUE;
+ }
+ session->media.has_video = 1;
+ session->media.remote_video_rtp_port = m->port;
+ session->media.remote_video_rtcp_port = m->port+1; /* FIXME We're assuming RTCP is on the next port */
+ if(m->direction == JANUS_SDP_SENDONLY || m->direction == JANUS_SDP_INACTIVE)
+ session->media.video_send = FALSE;
+ else
+ session->media.video_send = TRUE;
+ } else {
+ session->media.video_send = FALSE;
+ }
+ } else {
+ JANUS_LOG(LOG_WARN, "Unsupported media line (not audio/video)\n");
+ temp = temp->next;
+ continue;
+ }
+ if(m->c_addr) {
+ if(update && strcmp(m->c_addr, session->media.remote_ip)) {
+ /* This is an update and an address changed */
+ if(changed)
+ *changed = TRUE;
+ }
+ g_free(session->media.remote_ip);
+ session->media.remote_ip = g_strdup(m->c_addr);
+ }
+ if(update) {
+ /* FIXME This is a session update, we only accept changes in IP/ports */
+ temp = temp->next;
+ continue;
+ }
+ GList *tempA = m->attributes;
+ while(tempA) {
+ janus_sdp_attribute *a = (janus_sdp_attribute *)tempA->data;
+ if(a->name) {
+ if(!strcasecmp(a->name, "crypto")) {
+ if(m->type == JANUS_SDP_AUDIO || m->type == JANUS_SDP_VIDEO) {
+ gint32 tag = 0;
+ int suite;
+ char crypto[81];
+ /* FIXME inline can be more complex than that, and we're currently only offering SHA1_80 */
+ int res = sscanf(a->value, "%"SCNi32" AES_CM_128_HMAC_SHA1_%2d inline:%80s",
+ &tag, &suite, crypto);
+ if(res != 3) {
+ JANUS_LOG(LOG_WARN, "Failed to parse crypto line, ignoring... %s\n", a->value);
+ } else {
+ gboolean video = (m->type == JANUS_SDP_VIDEO);
+ int current_suite = video ? session->media.video_srtp_suite_in : session->media.audio_srtp_suite_in;
+ if(current_suite == 0) {
+ if(video)
+ session->media.video_srtp_suite_in = suite;
+ else
+ session->media.audio_srtp_suite_in = suite;
+ janus_sipre_srtp_set_remote(session, video, crypto, suite);
+ session->media.has_srtp_remote = TRUE;
+ } else {
+ JANUS_LOG(LOG_WARN, "We already configured a %s crypto context (AES_CM_128_HMAC_SHA1_%d), skipping additional crypto line\n",
+ video ? "video" : "audio", current_suite);
+ }
+ }
+ }
+ }
+ }
+ tempA = tempA->next;
+ }
+ if(answer && (m->type == JANUS_SDP_AUDIO || m->type == JANUS_SDP_VIDEO)) {
+ /* Check which codec was negotiated eventually */
+ int pt = -1;
+ if(m->ptypes)
+ pt = GPOINTER_TO_INT(m->ptypes->data);
+ if(pt > -1) {
+ if(m->type == JANUS_SDP_AUDIO) {
+ session->media.audio_pt = pt;
+ } else {
+ session->media.video_pt = pt;
+ }
+ }
+ }
+ temp = temp->next;
+ }
+ if(update && changed && *changed) {
+ /* Something changed: mark this on the session, so that the thread can update the sockets */
+ session->media.updated = TRUE;
+ if(session->media.pipefd[1] > 0) {
+ int code = 1;
+ ssize_t res = 0;
+ do {
+ res = write(session->media.pipefd[1], &code, sizeof(int));
+ } while(res == -1 && errno == EINTR);
+ }
+ }
+}
+
+char *janus_sipre_sdp_manipulate(janus_sipre_session *session, janus_sdp *sdp, gboolean answer) {
+ if(!session || !sdp)
+ return NULL;
+ /* Start replacing stuff */
+ JANUS_LOG(LOG_VERB, "Setting protocol to %s\n", session->media.require_srtp ? "RTP/SAVP" : "RTP/AVP");
+ GList *temp = sdp->m_lines;
+ while(temp) {
+ janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
+ g_free(m->proto);
+ m->proto = g_strdup(session->media.require_srtp ? "RTP/SAVP" : "RTP/AVP");
+ if(m->type == JANUS_SDP_AUDIO) {
+ m->port = session->media.local_audio_rtp_port;
+ if(session->media.has_srtp_local) {
+ char *crypto = NULL;
+ session->media.audio_srtp_suite_out = 80;
+ janus_sipre_srtp_set_local(session, FALSE, &crypto);
+ /* FIXME 32? 80? Both? */
+ janus_sdp_attribute *a = janus_sdp_attribute_create("crypto", "1 AES_CM_128_HMAC_SHA1_80 inline:%s", crypto);
+ g_free(crypto);
+ m->attributes = g_list_append(m->attributes, a);
+ }
+ } else if(m->type == JANUS_SDP_VIDEO) {
+ m->port = session->media.local_video_rtp_port;
+ if(session->media.has_srtp_local) {
+ char *crypto = NULL;
+ session->media.audio_srtp_suite_out = 80;
+ janus_sipre_srtp_set_local(session, TRUE, &crypto);
+ /* FIXME 32? 80? Both? */
+ janus_sdp_attribute *a = janus_sdp_attribute_create("crypto", "1 AES_CM_128_HMAC_SHA1_80 inline:%s", crypto);
+ g_free(crypto);
+ m->attributes = g_list_append(m->attributes, a);
+ }
+ }
+ g_free(m->c_addr);
+ m->c_addr = g_strdup(local_ip);
+ if(answer && (m->type == JANUS_SDP_AUDIO || m->type == JANUS_SDP_VIDEO)) {
+ /* Check which codec was negotiated eventually */
+ int pt = -1;
+ if(m->ptypes)
+ pt = GPOINTER_TO_INT(m->ptypes->data);
+ if(pt > -1) {
+ if(m->type == JANUS_SDP_AUDIO) {
+ session->media.audio_pt = pt;
+ } else {
+ session->media.video_pt = pt;
+ }
+ }
+ }
+ temp = temp->next;
+ }
+ /* Generate a SDP string out of our changes */
+ return janus_sdp_write(sdp);
+}
+
+/* Bind local RTP/RTCP sockets */
+static int janus_sipre_allocate_local_ports(janus_sipre_session *session) {
+ if(session == NULL) {
+ JANUS_LOG(LOG_ERR, "Invalid session\n");
+ return -1;
+ }
+ /* Reset status */
+ if(session->media.audio_rtp_fd != -1) {
+ close(session->media.audio_rtp_fd);
+ session->media.audio_rtp_fd = -1;
+ }
+ if(session->media.audio_rtcp_fd != -1) {
+ close(session->media.audio_rtcp_fd);
+ session->media.audio_rtcp_fd = -1;
+ }
+ session->media.local_audio_rtp_port = 0;
+ session->media.local_audio_rtcp_port = 0;
+ session->media.audio_ssrc = 0;
+ if(session->media.video_rtp_fd != -1) {
+ close(session->media.video_rtp_fd);
+ session->media.video_rtp_fd = -1;
+ }
+ if(session->media.video_rtcp_fd != -1) {
+ close(session->media.video_rtcp_fd);
+ session->media.video_rtcp_fd = -1;
+ }
+ session->media.local_video_rtp_port = 0;
+ session->media.local_video_rtcp_port = 0;
+ session->media.video_ssrc = 0;
+ if(session->media.pipefd[0] > 0) {
+ close(session->media.pipefd[0]);
+ session->media.pipefd[0] = -1;
+ }
+ if(session->media.pipefd[1] > 0) {
+ close(session->media.pipefd[1]);
+ session->media.pipefd[1] = -1;
+ }
+ /* Start */
+ int attempts = 100; /* FIXME Don't retry forever */
+ if(session->media.has_audio) {
+ JANUS_LOG(LOG_VERB, "Allocating audio ports:\n");
+ struct sockaddr_in audio_rtp_address, audio_rtcp_address;
+ while(session->media.local_audio_rtp_port == 0 || session->media.local_audio_rtcp_port == 0) {
+ if(attempts == 0) /* Too many failures */
+ return -1;
+ if(session->media.audio_rtp_fd == -1) {
+ session->media.audio_rtp_fd = socket(AF_INET, SOCK_DGRAM, 0);
+ }
+ if(session->media.audio_rtcp_fd == -1) {
+ session->media.audio_rtcp_fd = socket(AF_INET, SOCK_DGRAM, 0);
+ }
+ int rtp_port = g_random_int_range(10000, 60000); /* FIXME Should this be configurable? */
+ if(rtp_port % 2)
+ rtp_port++; /* Pick an even port for RTP */
+ audio_rtp_address.sin_family = AF_INET;
+ audio_rtp_address.sin_port = htons(rtp_port);
+ inet_pton(AF_INET, local_ip, &audio_rtp_address.sin_addr.s_addr);
+ if(bind(session->media.audio_rtp_fd, (struct sockaddr *)(&audio_rtp_address), sizeof(struct sockaddr)) < 0) {
+ JANUS_LOG(LOG_ERR, "Bind failed for audio RTP (port %d), trying a different one...\n", rtp_port);
+ attempts--;
+ continue;
+ }
+ JANUS_LOG(LOG_VERB, "Audio RTP listener bound to port %d\n", rtp_port);
+ int rtcp_port = rtp_port+1;
+ audio_rtcp_address.sin_family = AF_INET;
+ audio_rtcp_address.sin_port = htons(rtcp_port);
+ inet_pton(AF_INET, local_ip, &audio_rtcp_address.sin_addr.s_addr);
+ if(bind(session->media.audio_rtcp_fd, (struct sockaddr *)(&audio_rtcp_address), sizeof(struct sockaddr)) < 0) {
+ JANUS_LOG(LOG_ERR, "Bind failed for audio RTCP (port %d), trying a different one...\n", rtcp_port);
+ /* RTP socket is not valid anymore, reset it */
+ close(session->media.audio_rtp_fd);
+ session->media.audio_rtp_fd = -1;
+ attempts--;
+ continue;
+ }
+ JANUS_LOG(LOG_VERB, "Audio RTCP listener bound to port %d\n", rtcp_port);
+ session->media.local_audio_rtp_port = rtp_port;
+ session->media.local_audio_rtcp_port = rtcp_port;
+ }
+ }
+ if(session->media.has_video) {
+ JANUS_LOG(LOG_VERB, "Allocating video ports:\n");
+ struct sockaddr_in video_rtp_address, video_rtcp_address;
+ while(session->media.local_video_rtp_port == 0 || session->media.local_video_rtcp_port == 0) {
+ if(attempts == 0) /* Too many failures */
+ return -1;
+ if(session->media.video_rtp_fd == -1) {
+ session->media.video_rtp_fd = socket(AF_INET, SOCK_DGRAM, 0);
+ }
+ if(session->media.video_rtcp_fd == -1) {
+ session->media.video_rtcp_fd = socket(AF_INET, SOCK_DGRAM, 0);
+ }
+ int rtp_port = g_random_int_range(10000, 60000); /* FIXME Should this be configurable? */
+ if(rtp_port % 2)
+ rtp_port++; /* Pick an even port for RTP */
+ video_rtp_address.sin_family = AF_INET;
+ video_rtp_address.sin_port = htons(rtp_port);
+ inet_pton(AF_INET, local_ip, &video_rtp_address.sin_addr.s_addr);
+ if(bind(session->media.video_rtp_fd, (struct sockaddr *)(&video_rtp_address), sizeof(struct sockaddr)) < 0) {
+ JANUS_LOG(LOG_ERR, "Bind failed for video RTP (port %d), trying a different one...\n", rtp_port);
+ attempts--;
+ continue;
+ }
+ JANUS_LOG(LOG_VERB, "Video RTP listener bound to port %d\n", rtp_port);
+ int rtcp_port = rtp_port+1;
+ video_rtcp_address.sin_family = AF_INET;
+ video_rtcp_address.sin_port = htons(rtcp_port);
+ inet_pton(AF_INET, local_ip, &video_rtcp_address.sin_addr.s_addr);
+ if(bind(session->media.video_rtcp_fd, (struct sockaddr *)(&video_rtcp_address), sizeof(struct sockaddr)) < 0) {
+ JANUS_LOG(LOG_ERR, "Bind failed for video RTCP (port %d), trying a different one...\n", rtcp_port);
+ /* RTP socket is not valid anymore, reset it */
+ close(session->media.video_rtp_fd);
+ session->media.video_rtp_fd = -1;
+ attempts--;
+ continue;
+ }
+ JANUS_LOG(LOG_VERB, "Video RTCP listener bound to port %d\n", rtcp_port);
+ session->media.local_video_rtp_port = rtp_port;
+ session->media.local_video_rtcp_port = rtcp_port;
+ }
+ }
+ /* We need this to quickly interrupt the poll when it's time to update a session or wrap up */
+ pipe(session->media.pipefd);
+ return 0;
+}
+
+/* Helper method to (re)connect RTP/RTCP sockets */
+static void janus_sipre_connect_sockets(janus_sipre_session *session, struct sockaddr_in *server_addr) {
+ if(!session || !server_addr)
+ return;
+
+ if(session->media.updated) {
+ JANUS_LOG(LOG_VERB, "Updating session sockets\n");
+ }
+
+ /* Connect peers (FIXME This pretty much sucks right now) */
+ if(session->media.remote_audio_rtp_port) {
+ server_addr->sin_port = htons(session->media.remote_audio_rtp_port);
+ if(connect(session->media.audio_rtp_fd, (struct sockaddr *)server_addr, sizeof(struct sockaddr)) == -1) {
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] Couldn't connect audio RTP? (%s:%d)\n", session->account.username, session->media.remote_ip, session->media.remote_audio_rtp_port);
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] -- %d (%s)\n", session->account.username, errno, strerror(errno));
+ }
+ }
+ if(session->media.remote_audio_rtcp_port) {
+ server_addr->sin_port = htons(session->media.remote_audio_rtcp_port);
+ if(connect(session->media.audio_rtcp_fd, (struct sockaddr *)server_addr, sizeof(struct sockaddr)) == -1) {
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] Couldn't connect audio RTCP? (%s:%d)\n", session->account.username, session->media.remote_ip, session->media.remote_audio_rtcp_port);
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] -- %d (%s)\n", session->account.username, errno, strerror(errno));
+ }
+ }
+ if(session->media.remote_video_rtp_port) {
+ server_addr->sin_port = htons(session->media.remote_video_rtp_port);
+ if(connect(session->media.video_rtp_fd, (struct sockaddr *)server_addr, sizeof(struct sockaddr)) == -1) {
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] Couldn't connect video RTP? (%s:%d)\n", session->account.username, session->media.remote_ip, session->media.remote_video_rtp_port);
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] -- %d (%s)\n", session->account.username, errno, strerror(errno));
+ }
+ }
+ if(session->media.remote_video_rtcp_port) {
+ server_addr->sin_port = htons(session->media.remote_video_rtcp_port);
+ if(connect(session->media.video_rtcp_fd, (struct sockaddr *)server_addr, sizeof(struct sockaddr)) == -1) {
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] Couldn't connect video RTCP? (%s:%d)\n", session->account.username, session->media.remote_ip, session->media.remote_video_rtcp_port);
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] -- %d (%s)\n", session->account.username, errno, strerror(errno));
+ }
+ }
+
+}
+
+/* Thread to relay RTP/RTCP frames coming from the SIPre peer */
+static void *janus_sipre_relay_thread(void *data) {
+ janus_sipre_session *session = (janus_sipre_session *)data;
+ if(!session || !session->account.username || !session->callee) {
+ g_thread_unref(g_thread_self());
+ return NULL;
+ }
+ JANUS_LOG(LOG_VERB, "Starting relay thread (%s <--> %s)\n", session->account.username, session->callee);
+
+ gboolean have_server_ip = TRUE;
+ struct sockaddr_in server_addr;
+ memset(&server_addr, 0, sizeof(server_addr));
+ server_addr.sin_family = AF_INET;
+ if((inet_aton(session->media.remote_ip, &server_addr.sin_addr)) <= 0) { /* Not a numeric IP... */
+ struct hostent *host = gethostbyname(session->media.remote_ip); /* ...resolve name */
+ if(!host) {
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] Couldn't get host (%s)\n", session->account.username, session->media.remote_ip);
+ have_server_ip = FALSE;
+ } else {
+ server_addr.sin_addr = *(struct in_addr *)host->h_addr_list;
+ }
+ }
+ if(have_server_ip)
+ janus_sipre_connect_sockets(session, &server_addr);
+
+ if(!session->callee) {
+ JANUS_LOG(LOG_VERB, "[SIPre-%s] Leaving thread, no callee...\n", session->account.username);
+ g_thread_unref(g_thread_self());
+ return NULL;
+ }
+ /* File descriptors */
+ socklen_t addrlen;
+ struct sockaddr_in remote;
+ int resfd = 0, bytes = 0;
+ struct pollfd fds[5];
+ int pipe_fd = session->media.pipefd[0];
+ char buffer[1500];
+ memset(buffer, 0, 1500);
+ /* Loop */
+ int num = 0;
+ gboolean goon = TRUE;
+ int astep = 0, vstep = 0;
+ guint32 ats = 0, vts = 0;
+ while(goon && session != NULL && !session->destroyed &&
+ session->status > janus_sipre_call_status_idle &&
+ session->status < janus_sipre_call_status_closing) { /* FIXME We need a per-call watchdog as well */
+
+ if(session->media.updated) {
+ /* Apparently there was a session update */
+ if(have_server_ip && (inet_aton(session->media.remote_ip, &server_addr.sin_addr)) <= 0) {
+ janus_sipre_connect_sockets(session, &server_addr);
+ } else {
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] Couldn't update session details (missing or invalid remote IP address)\n", session->account.username);
+ }
+ session->media.updated = FALSE;
+ }
+
+ /* Prepare poll */
+ num = 0;
+ if(session->media.audio_rtp_fd != -1) {
+ fds[num].fd = session->media.audio_rtp_fd;
+ fds[num].events = POLLIN;
+ fds[num].revents = 0;
+ num++;
+ }
+ if(session->media.audio_rtcp_fd != -1) {
+ fds[num].fd = session->media.audio_rtcp_fd;
+ fds[num].events = POLLIN;
+ fds[num].revents = 0;
+ num++;
+ }
+ if(session->media.video_rtp_fd != -1) {
+ fds[num].fd = session->media.video_rtp_fd;
+ fds[num].events = POLLIN;
+ fds[num].revents = 0;
+ num++;
+ }
+ if(session->media.video_rtcp_fd != -1) {
+ fds[num].fd = session->media.video_rtcp_fd;
+ fds[num].events = POLLIN;
+ fds[num].revents = 0;
+ num++;
+ }
+ if(pipe_fd != -1) {
+ fds[num].fd = pipe_fd;
+ fds[num].events = POLLIN;
+ fds[num].revents = 0;
+ num++;
+ }
+ /* Wait for some data */
+ resfd = poll(fds, num, 1000);
+ if(resfd < 0) {
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] Error polling...\n", session->account.username);
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] -- %d (%s)\n", session->account.username, errno, strerror(errno));
+ break;
+ } else if(resfd == 0) {
+ /* No data, keep going */
+ continue;
+ }
+ if(session == NULL || session->destroyed ||
+ session->status <= janus_sipre_call_status_idle ||
+ session->status >= janus_sipre_call_status_closing)
+ break;
+ int i = 0;
+ for(i=0; i<num; i++) {
+ if(fds[i].revents & (POLLERR | POLLHUP)) {
+ /* Socket error? */
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] Error polling: %s...\n", session->account.username,
+ fds[i].revents & POLLERR ? "POLLERR" : "POLLHUP");
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] -- %d (%s)\n", session->account.username, errno, strerror(errno));
+ if(session->media.updated)
+ break;
+ goon = FALSE; /* Can we assume it's pretty much over, after a POLLERR? */
+ /* FIXME Simulate a "hangup" coming from the browser */
+ janus_sipre_message *msg = g_malloc0(sizeof(janus_sipre_message));
+ msg->handle = session->handle;
+ msg->message = json_pack("{ss}", "request", "hangup");
+ msg->transaction = NULL;
+ msg->jsep = NULL;
+ g_async_queue_push(messages, msg);
+ break;
+ } else if(fds[i].revents & POLLIN) {
+ if(pipe_fd != -1 && fds[i].fd == pipe_fd) {
+ /* Poll interrupted for a reason, go on */
+ int code = 0;
+ bytes = read(pipe_fd, &code, sizeof(int));
+ break;
+ }
+ /* Got an RTP/RTCP packet */
+ addrlen = sizeof(remote);
+ bytes = recvfrom(fds[i].fd, buffer, 1500, 0, (struct sockaddr*)&remote, &addrlen);
+ /* Let's check what this is */
+ gboolean video = fds[i].fd == session->media.video_rtp_fd || fds[i].fd == session->media.video_rtcp_fd;
+ gboolean rtcp = fds[i].fd == session->media.audio_rtcp_fd || fds[i].fd == session->media.video_rtcp_fd;
+ if(!rtcp) {
+ /* Audio or Video RTP */
+ rtp_header *header = (rtp_header *)buffer;
+ if((video && session->media.video_ssrc_peer != ntohl(header->ssrc)) ||
+ (!video && session->media.audio_ssrc_peer != ntohl(header->ssrc))) {
+ if(video) {
+ session->media.video_ssrc_peer = ntohl(header->ssrc);
+ } else {
+ session->media.audio_ssrc_peer = ntohl(header->ssrc);
+ }
+ JANUS_LOG(LOG_VERB, "[SIPre-%s] Got SIP peer %s SSRC: %"SCNu32"\n",
+ session->account.username ? session->account.username : "unknown",
+ video ? "video" : "audio", session->media.audio_ssrc_peer);
+ }
+ /* Is this SRTP? */
+ if(session->media.has_srtp_remote) {
+ int buflen = bytes;
+ srtp_err_status_t res = srtp_unprotect(
+ (video ? session->media.video_srtp_in : session->media.audio_srtp_in),
+ buffer, &buflen);
+ if(res != srtp_err_status_ok && res != srtp_err_status_replay_fail && res != srtp_err_status_replay_old) {
+ guint32 timestamp = ntohl(header->timestamp);
+ guint16 seq = ntohs(header->seq_number);
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] %s SRTP unprotect error: %s (len=%d-->%d, ts=%"SCNu32", seq=%"SCNu16")\n",
+ session->account.username ? session->account.username : "unknown",
+ video ? "Video" : "Audio", janus_srtp_error_str(res), bytes, buflen, timestamp, seq);
+ continue;
+ }
+ bytes = buflen;
+ }
+ /* Check if the SSRC changed (e.g., after a re-INVITE or UPDATE) */
+ guint32 timestamp = ntohl(header->timestamp);
+ janus_rtp_header_update(header, &session->media.context, video,
+ (video ? (vstep ? vstep : 4500) : (astep ? astep : 960)));
+ if(video) {
+ if(vts == 0) {
+ vts = timestamp;
+ } else if(vstep == 0) {
+ vstep = timestamp-vts;
+ if(vstep < 0) {
+ vstep = 0;
+ }
+ }
+ } else {
+ if(ats == 0) {
+ ats = timestamp;
+ } else if(astep == 0) {
+ astep = timestamp-ats;
+ if(astep < 0) {
+ astep = 0;
+ }
+ }
+ }
+ /* Save the frame if we're recording */
+ janus_recorder_save_frame(video ? session->vrc_peer : session->arc_peer, buffer, bytes);
+ /* Relay to browser */
+ gateway->relay_rtp(session->handle, video, buffer, bytes);
+ continue;
+ } else {
+ /* Audio or Video RTCP */
+ if(session->media.has_srtp_remote) {
+ int buflen = bytes;
+ srtp_err_status_t res = srtp_unprotect_rtcp(
+ (video ? session->media.video_srtp_in : session->media.audio_srtp_in),
+ buffer, &buflen);
+ if(res != srtp_err_status_ok && res != srtp_err_status_replay_fail && res != srtp_err_status_replay_old) {
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] %s SRTCP unprotect error: %s (len=%d-->%d)\n",
+ session->account.username ? session->account.username : "unknown",
+ video ? "Video" : "Audio", janus_srtp_error_str(res), bytes, buflen);
+ continue;
+ }
+ bytes = buflen;
+ }
+ /* Relay to browser */
+ gateway->relay_rtcp(session->handle, video, buffer, bytes);
+ continue;
+ }
+ }
+ }
+ }
+ if(session->media.audio_rtp_fd != -1) {
+ close(session->media.audio_rtp_fd);
+ session->media.audio_rtp_fd = -1;
+ }
+ if(session->media.audio_rtcp_fd != -1) {
+ close(session->media.audio_rtcp_fd);
+ session->media.audio_rtcp_fd = -1;
+ }
+ session->media.local_audio_rtp_port = 0;
+ session->media.local_audio_rtcp_port = 0;
+ session->media.audio_ssrc = 0;
+ if(session->media.video_rtp_fd != -1) {
+ close(session->media.video_rtp_fd);
+ session->media.video_rtp_fd = -1;
+ }
+ if(session->media.video_rtcp_fd != -1) {
+ close(session->media.video_rtcp_fd);
+ session->media.video_rtcp_fd = -1;
+ }
+ session->media.local_video_rtp_port = 0;
+ session->media.local_video_rtcp_port = 0;
+ session->media.video_ssrc = 0;
+ if(session->media.pipefd[0] > 0) {
+ close(session->media.pipefd[0]);
+ session->media.pipefd[0] = -1;
+ }
+ if(session->media.pipefd[1] > 0) {
+ close(session->media.pipefd[1]);
+ session->media.pipefd[1] = -1;
+ }
+ /* Clean up SRTP stuff, if needed */
+ janus_sipre_srtp_cleanup(session);
+ /* Done */
+ JANUS_LOG(LOG_VERB, "Leaving SIPre relay thread\n");
+ g_thread_unref(g_thread_self());
+ return NULL;
+}
+
+
+/* libre loop thread */
+gpointer janus_sipre_stack_thread(gpointer user_data) {
+ JANUS_LOG(LOG_FATAL, "Joining libre loop thread...\n");
+ /* Enter loop */
+ int err = re_main(NULL);
+ if(err != 0) {
+ JANUS_LOG(LOG_ERR, "re_main() failed: %d (%s)\n", err, strerror(err));
+ }
+ JANUS_LOG(LOG_FATAL, "Leaving libre loop thread...\n");
+ g_thread_unref(g_thread_self());
+ return NULL;
+}
+
+/* Called when challenged for credentials */
+int janus_sipre_cb_auth(char **user, char **pass, const char *realm, void *arg) {
+ janus_sipre_session *session = (janus_sipre_session *)arg;
+ JANUS_LOG(LOG_INFO, "[SIPre-%s] janus_sipre_cb_auth (realm=%s)\n", session->account.username, realm);
+ /* TODO How do we handle hashed secrets? */
+ int err = 0;
+ err |= str_dup(user, session->account.authuser);
+ err |= str_dup(pass, session->account.secret);
+ JANUS_LOG(LOG_INFO, "[SIPre-%s] -- %s / %s\n", session->account.username, *user, *pass);
+ return err;
+}
+
+/* Called when REGISTER responses are received */
+void janus_sipre_cb_register(int err, const struct sip_msg *msg, void *arg) {
+ janus_sipre_session *session = (janus_sipre_session *)arg;
+ JANUS_LOG(LOG_INFO, "[SIPre-%s] janus_sipre_cb_register\n", session->account.username);
+ if(err) {
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] REGISTER error: %s\n", session->account.username, strerror(err));
+ } else {
+ JANUS_LOG(LOG_INFO, "[SIPre-%s] REGISTER reply: %u %s\n", session->account.username, msg->scode, (char *)&msg->reason.p);
+ }
+ /* TODO Send result back to user */
+}
+
+/* Called when SIP progress (e.g., 180 Ringing) responses are received */
+void janus_sipre_cb_progress(const struct sip_msg *msg, void *arg) {
+ janus_sipre_session *session = (janus_sipre_session *)arg;
+ JANUS_LOG(LOG_INFO, "[SIPre-%s] session progress: %u %s\n", session->account.username, msg->scode, (char *)&msg->reason.p);
+
+ /* TODO Handle */
+}
+
+/* Called upon incoming INVITEs */
+void janus_sipre_cb_incoming(const struct sip_msg *msg, void *arg) {
+ janus_sipre_session *session = (janus_sipre_session *)arg;
+ JANUS_LOG(LOG_INFO, "[SIPre-%s] janus_sipre_cb_incoming\n", session->account.username);
+
+ /* TODO Handle */
+}
+
+/* Called when an SDP offer is received (got offer: true) or being sent (got_offer: false) */
+int janus_sipre_cb_offer(struct mbuf **mbp, const struct sip_msg *msg, void *arg) {
+ janus_sipre_session *session = (janus_sipre_session *)arg;
+ JANUS_LOG(LOG_INFO, "[SIPre-%s] janus_sipre_cb_offer\n", session->account.username);
+
+ struct sdp_session *sdp = NULL;
+ const bool got_offer = mbuf_get_left(msg->mb);
+ if(got_offer) {
+ int err = sdp_decode(sdp, msg->mb, true);
+ if(err) {
+ JANUS_LOG(LOG_ERR, "unable to decode SDP offer: %s\n", strerror(err));
+ return err;
+ }
+ JANUS_LOG(LOG_INFO, "SDP offer received\n");
+ /* TODO Handle */
+ } else {
+ JANUS_LOG(LOG_INFO, "sending SDP offer\n");
+ }
+
+ return sdp_encode(mbp, sdp, !got_offer);
+}
+
+
+/* called when an SDP answer is received */
+int janus_sipre_cb_answer(const struct sip_msg *msg, void *arg) {
+ janus_sipre_session *session = (janus_sipre_session *)arg;
+ JANUS_LOG(LOG_INFO, "[SIPre-%s] janus_sipre_cb_answer\n", session->account.username);
+
+ JANUS_LOG(LOG_INFO, "SDP answer received\n");
+
+ struct sdp_session *sdp = NULL;
+ int err = sdp_decode(sdp, msg->mb, false);
+ if(err) {
+ JANUS_LOG(LOG_ERR, "unable to decode SDP answer: %s\n", strerror(err));
+ return err;
+ }
+
+ /* TODO Handle */
+ return 0;
+}
+
+/* called when the session is established */
+void janus_sipre_cb_established(const struct sip_msg *msg, void *arg) {
+ janus_sipre_session *session = (janus_sipre_session *)arg;
+ JANUS_LOG(LOG_INFO, "[SIPre-%s] janus_sipre_cb_established\n", session->account.username);
+
+ /* TODO Handle */
+}
+
+/* Called when the session fails to connect or is terminated by the peer */
+void janus_sipre_cb_closed(int err, const struct sip_msg *msg, void *arg) {
+ janus_sipre_session *session = (janus_sipre_session *)arg;
+
+ if(err) {
+ JANUS_LOG(LOG_ERR, "[SIPre-%s] janus_sipre_cb_closed: %s\n", session->account.username, strerror(err));
+ } else {
+ JANUS_LOG(LOG_INFO, "[SIPre-%s] janus_sipre_cb_closed: %u %s\n", session->account.username, msg->scode, (char *)&msg->reason.p);
+ }
+
+ /* TODO Handle */
+}
+
+/* Called when all SIP transactions are completed */
+void janus_sipre_cb_exit(void *arg) {
+ /* Stop libre main loop */
+ re_cancel();
+}
diff --git a/utils.h b/utils.h
index b790418..dcafd4c 100644
--- a/utils.h
+++ b/utils.h
@@ -17,11 +17,14 @@
#include <netinet/in.h>
#include <jansson.h>
+#define JANUS_JSON_STRING JSON_STRING
+#define JANUS_JSON_INTEGER JSON_INTEGER
+#define JANUS_JSON_OBJECT JSON_OBJECT
/* Use JANUS_JSON_BOOL instead of the non-existing JSON_BOOLEAN */
-#define JANUS_JSON_BOOL JSON_TRUE
-#define JANUS_JSON_PARAM_REQUIRED 1
-#define JANUS_JSON_PARAM_POSITIVE 2
-#define JANUS_JSON_PARAM_NONEMPTY 4
+#define JANUS_JSON_BOOL JSON_TRUE
+#define JANUS_JSON_PARAM_REQUIRED 1
+#define JANUS_JSON_PARAM_POSITIVE 2
+#define JANUS_JSON_PARAM_NONEMPTY 4
struct janus_json_parameter {
const gchar *name;
--
Alioth's /usr/local/bin/git-commit-notice on /srv/git.debian.org/git/pkg-voip/janus.git
More information about the Pkg-voip-commits
mailing list