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http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1



1.  Tutorial Overview

WebSocket is a protocol that provides full-duplex communication between web clients and servers over TCP connections. Using the WebSocket protocol, browsers can connect to web servers and exchange data, regardless the type or nature of the application protocol. RFC 7118 leveraged this protocol in order to allow browsers to make VoIP calls using the SIP protocol.

This document describes how to use OpenSIPS as the core component of a SIP platform that connects both SIP clients (over UDP, TCP or TLS) as well as browser based clients (using SIP over WebSockets). While OpenSIPS handles the SIP signalling part, media is handled by RTPengine, a high performance media proxy that is able to handle both RTP and SRTP media streams, as well as bridging between them.

This tutorial is inspired from

2.  Setup

2.1  RTPengine

Installation

The RTPengine consists of two main components: a kernel module used to efficiently route the RTP packets directly in kernel, and a daemon used to communicate with OpenSIPS. You can find more details here. Both components can be installed from debs (on Debian based systems) or directly from sources. Simply follow the official documentation to install RTPengine.

Usage

After installing the kernel module and the additional libraries, the rtpengine daemon has to be configured. This can be done from /etc/default/ngcp-rtpengine-daemon if installed from debs, or from the command line if the daemon is started manually. On systemd based OSes, Eric Tamme created some startup scripts.

The interesting parameters we are using are as follows:

  • -i: the listening interface for RTP/SRTP
  • -n: the listening IP and port that is used by OpenSIPS to communicate with the RTPengine (NOTE: the rtpengine module only works with the rtpengine NG protocol, so you must use -n/--listen-ng; Using -u/--listen-udp or -l/--listen-tcp will not work!)
  • -c: the IP and port of the CLI - this is used to gather statistics for the RTP/SRTP sessions
  • -m, -M: both take an integer as argument and together define the local port range from which rtpengine will allocate UDP ports for media traffic relay. Default to 30000 and 40000 respectively.
  • -L: indicates the debugging level

You can find all the parameters available here.

Here is an example that runs rtpengine from cli that talks with OpenSIPS over localhost and RTP using the 1.1.1.1 IP:

./rtpengine -p /var/run/rtpengine.pid -i eth0/1.1.1.1 -n 127.0.0.1:60000 -c 127.0.0.1:60001 -m 50000 -M 55000 -E -L 7
Troubleshoot

First make sure the rtpengine daemon is started:

ps -ef | grep rtpengine

If the rtpengine daemon does not start, make sure the xt_RTPENGINE kernel module is loaded:

lsmod | grep xt_RTPENGINE

If the module is not loaded, make sure the ip_tables and x_tables kernel modules are loaded. Also, check the logs for the last errors of the system

dmesg

2.2  OpenSIPS

In order to use WebSocket in OpenSIPS, one has to load the proto_ws into its configuration file and define a listener for the WebSocket protocol.

listen=ws:127.0.0.1:8080
...
loadmodule "proto_ws.so"

Next, the rtpengine module has to be loaded and configured to communicate with the rtpengine daemon.

loadmodule "rtpengine.so"
modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:60000")

Note that the rtpengine_sock parameter should be the same as the -n parameter sent to the rtpengine daemon, and OpenSIPS should have IP connectivity to that socket.

Next, the routing logic has to be changed in order to treat different the clients that use DTLS-SRTP, from the ones that use plain RTP and enable bridging if necessary. To do that, one can check if the request message was received over the WebSocket protocol. This can be achieved using the following code:

if (proto == WS)
    setflag(SRC_WS);

In case the request is a REGISTER, we want to store this information in the location table, so that we know then the user is called. To do that, we can set a branch flag before calling the save()function. This way, when the lookup() method returns, we will be able to determine whether the client uses WebSocket or not.

    if (is_method("REGISTER")) {
        if (isflagset(SRC_WS))
            setbflag(DST_WS);

        fix_nated_register();
        if (!save("location"))                                                                                                                                 
            sl_reply_error();

        exit;
    }

When a call is placed, based on the two flags (STR_WS and DST_WS) we can determine what our caller and callee can "speak" (either RTP or DTLS-SRTP) and instruct the rtpengine daemon how to handle the call. We can do that by tuning the parameters passed to the rtpengine_offer() function.

    if (isflagset(SRC_WS) && isbflagset(DST_WS))
        $var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
    else if (isflagset(SRC_WS) && !isbflagset(DST_WS))
        $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
    else if (!isflagset(SRC_WS) && isbflagset(DST_WS))
        $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
    else if (!isflagset(SRC_WS) && !isbflagset(DST_WS))
        $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";

    rtpengine_offer("$var(rtpengine_flags)");

The rtpengine_answer() function logic should look like this:

    if (isflagset(SRC_WS) && isbflagset(DST_WS))
        $var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
    else if (isflagset(SRC_WS) && !isbflagset(DST_WS))
        $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
    else if (!isflagset(SRC_WS) && isbflagset(DST_WS))
        $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
    else if (!isflagset(SRC_WS) && !isbflagset(DST_WS))
        $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";

    rtpengine_answer("$var(rtpengine_flags)");

Now, all we have to do is to close the RTP/SRTP session when the call is ended. To do that, we use the rtpengine_delete() function:

    if (is_method("BYE|CANCEL")) {                                                                                                                      
        rtpengine_delete();

Having done all these settings should provide a full setup for interconnecting SIP clients over both UDP, TCP, etc. protocols, as well as browser based SIP clients over WebSocket.

3.  Configuration file

The following configuration file is a minimal working example of a Residential script that can handle clients connections over both UDP and Websocket transports. This example assumes that the SDP offer is present in the INVITE from the UAC and the SDP answer is in the 200 OK from the UAS.

#
# OpenSIPS residential configuration script
#     by OpenSIPS Solutions <team@opensips-solutions.com>
#
# Please refer to the Core CookBook at:
#      http://www.opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#


####### Global Parameters #########

debug=3
log_stderror=no
log_facility=LOG_LOCAL0

fork=yes
children=4
auto_aliases=no

listen=udp:127.0.0.0:5060 # TODO: update with your local IP and port
listen=ws:127.0.0.0:8080 # TODO: update with your local IP and port

####### Modules Section ########

# set module path
mpath="/usr/local/lib/opensips/modules/"

#### SIGNALING module
loadmodule "signaling.so"

#### StateLess module
loadmodule "sl.so"

#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)

#### Record Route Module
loadmodule "rr.so"
modparam("rr", "append_fromtag", 0)

#### MAX ForWarD module
loadmodule "maxfwd.so"

#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"

#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)

#### URI module
loadmodule "uri.so"
modparam("uri", "use_uri_table", 0)

#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "db_mode",   0)

#### REGISTRAR module
loadmodule "registrar.so"

#### RTPengine protocol
loadmodule "rtpengine.so"
modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.0:60000")

#### Nathelper protocol
loadmodule "nathelper.so"
modparam("registrar|nathelper", "received_avp", "$avp(rcv)")

#### UDP protocol
loadmodule "proto_udp.so"

#### WebSocket protocol
loadmodule "proto_ws.so"


####### Routing Logic ########

# main request routing logic
route{
	if (!mf_process_maxfwd_header("10")) {
		sl_send_reply("483","Too Many Hops");
		exit;
	}

	if (has_totag()) {
		# sequential requests within a dialog should
		# take the path determined by record-routing
		if (loose_route()) {
			if (is_method("INVITE")) {
				# even if in most of the cases is useless, do RR for
				# re-INVITEs alos, as some buggy clients do change route set
				# during the dialog.
				record_route();
			}

			# route it out to whatever destination was set by loose_route()
			# in $du (destination URI).
			route(relay);
		} else {
			if ( is_method("ACK") ) {
				if ( t_check_trans() ) {
					# non loose-route, but stateful ACK; must be an ACK after
					# a 487 or e.g. 404 from upstream server
					t_relay();
					exit;
				} else {
					# ACK without matching transaction ->
					# ignore and discard
					exit;
				}
			}
			sl_send_reply("404","Not here");
		}
		exit;
	}

	# CANCEL processing
	if (is_method("CANCEL")) {
		if (t_check_trans())
			t_relay();
		exit;
	}

	t_check_trans();

	if (!is_method("REGISTER")) {
		if (from_uri!=myself) {
			# if caller is not local, then called number must be local
			if (!uri==myself) {
				send_reply("403","Rely forbidden");
				exit;
			}
		}
	}

	# preloaded route checking
	if (loose_route()) {
		xlog("L_ERR",
		"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
		if (!is_method("ACK"))
			sl_send_reply("403","Preload Route denied");
		exit;
	}

	# record routing
	if (!is_method("REGISTER|MESSAGE"))
		record_route();

	if (!uri==myself) {
		append_hf("P-hint: outbound\r\n");
		route(relay);
	}

	# requests for my domain
	if (is_method("PUBLISH|SUBSCRIBE")) {
		sl_send_reply("503", "Service Unavailable");
		exit;
	}

	# check if the clients are using WebSockets
	if (proto == WS)
		setflag(SRC_WS);

	# consider the client is behind NAT - always fix the contact
	fix_nated_contact();

	if (is_method("REGISTER")) {

		# indicate that the client supports DTLS
		# so we know when he is called
		if (isflagset(SRC_WS))
			setbflag(DST_WS);

		fix_nated_register();
		if (!save("location"))
			sl_reply_error();

		exit;
	}

	if ($rU==NULL) {
		# request with no Username in RURI
		sl_send_reply("484","Address Incomplete");
		exit;
	}

	# do lookup with method filtering
	if (!lookup("location","m")) {
		t_newtran();
		t_reply("404", "Not Found");
		exit;
	}

	route(relay);
}

route[relay] {
	# for INVITEs enable some additional helper routes
	if (is_method("INVITE")) {
		t_on_branch("handle_nat");
		t_on_reply("handle_nat");
	} else if (is_method("BYE|CANCEL")) {
		rtpengine_delete();
	}

	if (!t_relay()) {
		send_reply("500","Internal Error");
	};
	exit;
}

branch_route[handle_nat] {

	if (!is_method("INVITE") || !has_body("application/sdp"))
		return;

	if (isflagset(SRC_WS) && isbflagset(DST_WS))
		$var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
	else if (isflagset(SRC_WS) && !isbflagset(DST_WS))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
	else if (!isflagset(SRC_WS) && isbflagset(DST_WS))
		$var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
	else if (!isflagset(SRC_WS) && !isbflagset(DST_WS))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";

	rtpengine_offer("$var(rtpengine_flags)");
}

onreply_route[handle_nat] {

	fix_nated_contact();
	if (!has_body("application/sdp"))
		return;

	if (isflagset(SRC_WS) && isbflagset(DST_WS))
		$var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
	else if (isflagset(SRC_WS) && !isbflagset(DST_WS))
		$var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
	else if (!isflagset(SRC_WS) && isbflagset(DST_WS))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
	else if (!isflagset(SRC_WS) && !isbflagset(DST_WS))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";

	rtpengine_answer("$var(rtpengine_flags)");
}
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29 Multimedia Service Platform admin 2014-03-06 20344
28 How to install OpenSIPS on CentOS Debian etc admin 2014-03-05 21254
27 Opensips Installation, How to. admin 2014-03-05 17689
26 100% CPU usage opensips admin 2014-03-05 20596
25 A2Billing and OpenSIPS admin 2014-03-04 21522
24 Opensips_1.9 install guide this is great I like this admin 2014-03-04 27092
23 Opensips install debian admin 2014-03-03 21541
22 Open Source VOIP applications, both clients and servers. admin 2013-11-20 22080
21 OfficeSIP Server is freeware VoIP, SIP server for Windows admin 2013-09-11 23108
20 My new toy: Bluebox-ng admin 2013-04-06 36752
19 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 2013-04-06 35138
18 Asterisk Installation Asterisk Realtime configuration admin 2013-04-06 25765
17 The SIP Router Project admin 2013-04-06 24777
16 Kamailio :: A Quick Introduction admin 2013-04-06 22149
15 Welcome to the Smartvox Knowledgebase admin 2013-04-06 22599
14 Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database admin 2013-04-06 26884
13 OpenSIPS vs Asterisk admin 2013-04-06 57645
12 OpenSER_from_an_asterisk_POV file admin 2013-04-06 22556
11 Using SIP Devices behind NAT OPensip Asterisk IPPhone SIP Telephony file admin 2013-03-31 55235