    <chapter id="general">
	<title>General Information</title>
	<section id="aboutser">
	    <title>About <acronym>SIP</acronym> Express Router (<acronym>SER</acronym>)</title>
	    <para>
		SIP Express Router (<acronym>SER</acronym>) is an industrial-strength, free VoIP 
		server based on the Session Initiation Protocol (<acronym>SIP</acronym>, RFC3261). 
		It is engineered to power <acronym>IP</acronym> telephony infrastructures up to large 
		scale. The server keeps track of users, sets up VoIP sessions, 
		relays instant messages and creates space for new plug-in applications. 
		Its proven interoperability guarantees seamless integration with 
		components from other vendors, eliminating the risk of a single-vendor 
		trap. It has successfully participated in various interoperability 
		tests in which it worked with the products of other leading <acronym>SIP</acronym> vendors.
	    </para>
	    <para>
		The <acronym>SIP</acronym> Express Router enables a flexible plug-in model for new 
		applications: Third parties can easily link their plug-ins with 
		the server code and provide thereby advanced and customized services. 
		In this way, plug-ins such as RADIUS accounting,
		SMS gateway, ENUM queries, or presence agent have already been developed and are provided as 
		advanced features. Other modules are underway: 
		firewall control, postgres and LDAP database drivers and more.
	    </para>
	    <para>
		Its performance and robustness allows it to serve millions of users 
		and accommodate needs of very large operators. With a $3000 dual-CPU PC, the 
		<acronym>SIP</acronym> Express Router is able to power <acronym>IP</acronym> telephony services in an area 
		as large as the Bay Area during peak hours. Even on an IPAQ <acronym>PDA</acronym>, the server 
		withstands 150 calls per second (<acronym>CPS</acronym>)! The server has been powering our 
		iptel.org free <acronym>SIP</acronym> site withstanding heavy daily load that is further 
		increasing with the popularity of Microsoft's Windows Messenger.  
	    </para>
	    <para>
		The <acronym>SIP</acronym> Express Router is extremely configurable to allow the creation of 
		various routing and admission policies as well as setting up new and 
		customized services. Its configurability allows it to serve many roles: 
		network security barrier, application server, or <acronym>PSTN</acronym> gateway guard for 
		example.
	    </para>
	    <para>
		<application moreinfo="none">ser</application> can be also
		used with contributed applications. Currently, 
		<application moreinfo="none">serweb</application>, a
		<application moreinfo="none">ser</application> web interface,
		<application moreinfo="none">SIPSak</application> diagnostic tool 
		and 
		<application>SEMS</application> media server
		are available. Visit our site, 
		<ulink url="http://www.iptel.org/">http://www.iptel.org/</ulink>, 
                for more information on contributed packages.
	    </para>
	</section> 

	<section id="aboutiptel">
	    <title>About iptel.org</title>
	    <para>
		iptel.org is a know-how organization spun off from Germany's national 
		research company FhG Fokus. One of the first <acronym>SIP</acronym> implementations ever, 
		low-QoS enhancements, interoperability tests and VoIP-capable firewall 
		control concepts are examples of well-known FhG's work.
	    </para>
	    <para>
		iptel.org continues to keep this know-how leadership in <acronym>SIP</acronym>. 
		The access rate of the company's site, a well-known source of 
		technological information, is a best proof of interest. Thousands 
		of hits come every day from the whole Internet.
	    </para>
	    <para>
		The iptel.org site, powered by SER, offers SIP services on the public 
		Internet. Feel free to apply for a free SIP account at
		<ulink url="http://www.iptel.org/user/">http://www.iptel.org/user/
		</ulink>
	    </para>

	    
	</section> <!-- iptel -->
	<section id="serfeatures">
	    <title>Feature List</title>
	    <para>
		Based on the latest standards, the <acronym>SIP</acronym> Express Router (<acronym>SER</acronym>) includes 
		support for registrar, proxy and redirect mode. Further it acts as 
		an application server with support for instant messaging and 
		presence including a <acronym>2G/SMS</acronym> and Jabber gateway, a call control policy 
		language, call number translation, private dial plans and accounting, ENUM,
		authorization and authentication (<acronym>AAA</acronym>) services. <application>SER</application> runs on Sun/Solaris, 
		PC/Linux, PC/BSD, IPAQ/Linux platforms and supports  both <acronym>IPv4</acronym> and <acronym>IPv6</acronym>. 
		Hosting multiple domains and database redundancy is supported.
	    </para>
	    <para>
		<application>ser</application> has been carefully engineered with the following 
		design objectives in mind:
		<itemizedlist>
		    <listitem>
			<para>
			    <emphasis>Speed</emphasis> - With <application>ser</application>, 
			    thousands of calls per seconds are achievable 
			    even on low-cost platforms. This competitive capacity allows 
			    setting up networks which are inexpensive and easy to manage 
			    due to low number of devices required. The processing capacity 
			    makes dealing with many stress factors easier. The stress
			    factors may include but are not limited to broken configurations and implementations,
			    boot avalanches on power-up, high-traffic applications such as presence, 
			    redundancy replications and denial-of-service attacks.
			</para>
			<para> The speed has been achieved by extensive code optimization, use of customized code, 
			    <acronym>ANSI C</acronym> combined with assembly instructions and leveraging latest 
			    <acronym>SIP</acronym> improvements. When powered by a dual-CPU Linux PC, 
			    <application>ser</application> is able to process thousands of calls per second,
			    capacity needed to serve call signaling demands of Bay Area population.
			   
			</para>
		    </listitem>
		    <listitem>
			<para>
			    <emphasis>Flexibility</emphasis> - <application>SER</application> allows its users 
			    to define its behavior. 
			    Administrators may write textual scripts which determine <acronym>SIP</acronym> routing 
			    decisions, the main job of a proxy server. They may use the script to 
			    configure numerous parameters and introduce additional logic. For example, 
			    the scripts can determine for which destinations record routing should be 
			    performed, who will be authenticated, which transactions should be processed 
			    statefully, which requests will be proxied or redirected, etc.
			</para>

		    </listitem>
		    <listitem>
			<para>
			    <emphasis>Extensibility</emphasis> - <application>SER</application>'s extensibility allows linking of 
			    new C code to ser to 
			    redefine or extend its logic. The new code can be developed independently 
			    on <application>SER</application> core and linked to it in run-time. The concept is similar to 
			    the module concept known for example in Apache Web server. Even such essential parts such 
			    as transaction management have been developed as modules to keep the <application>SER</application> core 
			    compact and fast.
			</para>
		    </listitem>
		    <listitem>
			<para>
			    <emphasis>
				Portability.
			    </emphasis>
			    <application>ser</application> has been written in ANSI C. It has been extensively tested 
			    on PC/Linux and Sun/Solaris. Ports to BSD and IPAQ/Linux exist.
			</para>
		    </listitem>
		    <listitem>
			<para>
			    <emphasis>			   
				Interoperability. 
			</emphasis>
			<application>ser</application> is based on the open <acronym>SIP</acronym> standard. 
			    It has undergone extensive tests with products of other vendors both in iptel.org
			    labs and in the SIP Interoperability Tests (SIPIT). <application>ser</application> 
			    powers the public iptel.org site 24 hours a day, 356 days a year 
			    serving numerous SIP implementations using this site.
			</para>
		    </listitem>
		    <listitem>
			<para>
			    <emphasis>Small size.</emphasis>
			    Footprint of the core is 300k, add-on modules take up to 630k.
			</para>
		    </listitem>
		</itemizedlist>
	    </para>
	</section> <!-- serfeatures -->

	<section id="usecases">
	    <title>Use Cases</title>
	    <para>
		This section illustrates the most frequent uses of SIP. In all these scenarios, 
		the SIP Express Router (SER) can be easily deployed as the glue connecting all 
		SIP components together, be it soft-phones, hard-phones, PSTN gateways or any other 
		SIP-compliant devices.
	    </para>
	    <section>
		<title>Added-Value ISP Services</title>
		<para>
		    To attract customers, ISPs frequently offer applications bundled with IP access. 
		    With SIP, the providers can conveniently offer a variety of services running on top 
		    of a single infrastructure. Particularly, deploying VoIP and instant messaging and 
		    presence services is as easy as setting up a SIP server and guiding customers to use 
		    Windows Messenger. Additionally, the ISPs may offer advanced services such as PSTN 
		    termination, user-driven call handling or unified messaging all using the same infrastructure.
		</para>
		<para>
		    SIP Express Router has been engineered to power large scale networks: its capacity can deal 
		    with large number of customers under high load caused by modern applications. Premium 
		    performance allows deploying a low number of boxes while keeping investments and operational 
		    expenses extremely low. ISPs can offer SIP-based instant messaging services and interface
		    them to other instant messaging systems (Jabber, SMS). VoIP can be easily integrated along
		    with added-value services, such as voicemail.
		</para>
	    </section> <!-- Added-value ISP -->
	    <section>
		<title>PC2Phone</title>
		<para>
		    Internet Telephony Service Providers (ITSPs) offer the service of interconnecting 
		    Internet telephony users using PC softphone or appliances to PSTN. Particularly with 
		    long-distance and international calls, competitive pricing can be achieved by 
		    routing the calls over the Internet.
		</para>
		<para>
		    SIP Express Router can be easily configured to serve pc2phone users, distribute
		    calls to geographically appropriate PSTN gateway, act as a security barrier and keep 
		    track of charging. 
		</para>
	    </section>
	    <section>
		<title>PBX Replacement</title>
		<para>
		    Replacing a traditional PBX in an enterprise can achieve reasonable 
		    savings. Enterprises can deploy a single infrastructure for both voice 
		    and data and bridge distant locations over the Internet. Additionally, they can 
		    benefit of integration of voice and data.
		</para>
		<para>
		    The SIP Express Router scales from SOHOs to large, international enterprises. 
		    Even a single installation on a common PC is able to serve VoIP signaling of any 
		    world's enterprise. Its policy-based routing language makes implementation of numbering 
		    plans of companies spread across the world very easy. ACL features allow for protection of 
		    PSTN gateway from unauthorized callers.
		</para>
		<para>
		    SIP Express Router's support for programmable routing and accounting efficiently allows for 
		    implementation of such a scenario.
		</para>
	    </section>
	</section> <!-- Use Cases -->
	<section id="aboutsip">
	    <title>About SIP Technology</title>
	    <para>
		The SIP protocol family is the technology which integrates services. 
		With SIP, Internet users can easily contact each other; figure out 
		willingness to have a conversation and couple different applications such as VoIP, video 
		and instant messaging. Integration with added-value services is seamless and easy. Examples 
		include integration with web (click-to-dial), E-mail (voice2email, UMS), and PSTN-like 
		services (conditional forwarding).
	    </para>
	    <para>
		The core piece of the technology is the Session Initiation Protocol (SIP, RFC3261) standardized by IETF. 
		Its main function is to establish communication sessions between users connected to the public 
		Internet and identified by e-mail-like addresses. One of SIP's greatest features is its transparent 
		support for multiple applications: the same infrastructure may be used for voice, video, gaming 
		or instant messaging as well as any other communication application.
	    </para>
	    <para>
		There are numerous scenarios in which SIP is already deployed: PBX replacement allows for 
		deployment of single inexpensive infrastructure in enterprises; PC-2-phone long-distance 
		services (e.g., Deltathree) cut callers long-distance expenses; instant messaging offered 
		by public severs (e.g., iptel.org) combines voice and text services with presence information. 
		New deployment scenarios are underway: SIP is a part of UMTS networks and research publications 
		suggest the use of SIP for virtual home environments or distributed network games.
	    </para>
	</section> <!-- about sip -->
	<section id="serlimitations">
	    <title>Known SER Limitations</title>
	    <para>
		The following items are not part of current distribution and are planned for next releases:
		<itemizedlist>
		    <listitem>
			<para>
			    Script processing of multiple branches on forking
			</para>

			<warning>
			    <para>
				<application>ser</application>'s request processing language allows 
				to make request decisions based on current URI. 
				When a request if forked to multiple destinations, only the first branch's URI is used as 
				input for script processing. This might lead to unexpected results. Whenever a URI resolves 
				to multiple different next-hop URIs, only the first is processed which may result in handling 
				not appropriate for the other branch. For example, a URI might resolve to an IP phone SIP 
				address and PSTN gateway SIP address. If the IP phone address is the first, then script 
				execution ignores the second branch. If a script includes checking gateway address in
				request URI, the checks never match. That might result in ignoring of 
				gateway admission control rules or applying them unnecessarily to non-gateway 
				destinations.
			    </para>
			</warning>
		    </listitem>
		</itemizedlist>
	    </para>
	    <para>
		List of known problems is publicly available at the 
		<application>ser</application> webpage at
		<ulink url="http://www.iptel.org/ser/">
		    http://www.iptel.org/ser/
		</ulink>. See the "ISSUES" link.
	    </para>
	</section> <!-- limitations -->
	<section id="licensing">
	    <title>Licensing</title>
	    <para>
		<application>ser</application> is freely available under terms and
		conditions of the GNU General Public License.
	    </para>	
	    <!-- COPYING -->
	    <screen>
	    	    &gpllicense;
	    </screen>
	    
	</section>
	<section id="support">
	    <title>Obtaining Technical Assistance</title>
		<para>
			iptel.org offers qualified professional services. We help you to 
			plan your network, configure your server, build applications, 
			integrate SIP components with each other, and set up advanced features
			such as redundancy, multidomain support, CLID interworking and others
			not described in this document. Our customer alert services
			notifies you on all new features and code fixes. We help you to
			solve operational troubles in short time and keep you updated on
			latest operational practices. Ask info@iptel.org for
			information on enrollment in our support program.
		</para>

	    <para>
		Additionally, help may be obtained from our user forum. The community
		of <application>SER</application> users is subscribed to the
		serusers@iptel.org mailing list and discusses issues related to
		<application>SER</application> operation.
	    </para>
	    <itemizedlist>
		<title>Mailing List Instructions</title>
		<listitem>
		    <para>
			Public archives and subscription form:
			<ulink url="http://mail.iptel.org/mailman/listinfo/serusers">
			    http://mail.iptel.org/mailman/listinfo/serusers
			</ulink>			
		    </para>
		</listitem>
		<listitem>
		    <para>
			To post, send an email to serusers@iptel.org
		    </para>
		</listitem>
		<listitem>
		    <para>
			If you think you encountered an error, please submit the
			following information to avoid unnecessary round-trip times:			
			<itemizedlist>
			    <listitem>
				<para>
				    Name and version of your operating system --
				    you can obtain it by calling
				    <command>uname -a</command>
				</para>
			    </listitem>
			    <listitem>
				<para>
				    <application>ser</application> distribution: 
				    release number and package				    
				</para>
			    </listitem>
			    <listitem>
				<para>
				    <application>ser</application> build -- 
				    you can obtain it by calling 
				    <command moreinfo="none">ser -V</command>
				</para>
			    </listitem>
			    <listitem>
				<para>
				    Your <application>ser</application> configuration file
				</para>
			    </listitem>
			    <listitem>
				<para>
				    <application>ser</application> logs -- with default settings
				    few logs are printed to <command>syslog</command> facility which
				    typically dumps them to 
				    <filename moreinfo="none">/var/log/messages</filename>. To 
				    enable detailed logs dumped to <filename>stderr</filename>,
				    apply the following configuration options: <command moreinfo="none">
				    debug=8, log_stderror=yes, fork=no</command>.
				</para>
			    </listitem>
			    <listitem>
				<para>
				    Captured SIP messages -- you can obtain them 
				    using tools such as <application>ngrep</application>
				    or <application moreinfo="none">ethereal</application>.
				</para>
			    </listitem>
			</itemizedlist>
		    </para>
		    
		</listitem>
	    </itemizedlist>	    
	
	    <para>
		If you are concerned about your privacy and do not wish your
		queries to be posted and archived publicly, you may post to
		serhelp@iptel.org. E-mails to this address are only forwarded
		to iptel.org's <application>ser</application> development team.
		However, as the team is quite busy you should not be surprised
		to get replies with considerable delay.

	    </para>
	</section>
        
	<section id="moreinfo">
	    <title>More Information</title>
	    <para>
		Most up-to-date information including latest and most complete version
		of this documentation is always available at our website,
		<ulink url="http://www.iptel.org/ser/">http://www.iptel.org/ser/</ulink>.
		The site includes links to other important information about
		<application moreinfo="none">ser</application>, such
		as installation guidelines (INSTALL), download links,
		development pages, programmer's manual, etc.
	    </para>
	    <para>
		A SIP tutorial (slide set) is available at 
		<ulink url="http://www.iptel.org/sip/">http://www.iptel.org/sip/</ulink> .
	    </para>
	</section> <!-- info -->

	<section>
	    <title>Release Notes</title>
	    <literallayout format="linespecific" class="normal">
&releasenotes;
	    </literallayout>
	</section> <!-- release notes -->


    </chapter> <!-- general -->
