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SIP Express Router (SER) is an industrial-strength, free VoIP server based on the Session Initiation Protocol (SIP, RFC3261). It is engineered to power IP 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 SIP vendors. The SIP 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. 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 SIP Express Router is able to power IP telephony services in an area as large as the Bay Area during peak hours. Even on an IPAQ PDA, the server withstands 150 calls per second (CPS)! The server has been powering our iptel.org free SIP site withstanding heavy daily load that is further increasing with the popularity of Microsoft's Windows Messenger. The SIP 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 PSTN gateway guard for example. ser can be also used with contributed applications. Currently, serweb, a ser web interface, SIPSak diagnostic tool and SEMS media server are available. Visit our site, http://www.iptel.org/, for more information on contributed packages.

iptel.org is a know-how organization spun off from Germany's national research company FhG Fokus. One of the first SIP implementations ever, low-QoS enhancements, interoperability tests and VoIP-capable firewall control concepts are examples of well-known FhG's work. iptel.org continues to keep this know-how leadership in SIP. 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. The iptel.org site, powered by SER, offers SIP services on the public Internet. Feel free to apply for a free SIP account at http://www.iptel.org/user/

Based on the latest standards, the SIP Express Router (SER) includes support for registrar, proxy and redirect mode. Further it acts as an application server with support for instant messaging and presence including a 2G/SMS and Jabber gateway, a call control policy language, call number translation, private dial plans and accounting, ENUM, authorization and authentication (AAA) services. SER runs on Sun/Solaris, PC/Linux, PC/BSD, IPAQ/Linux platforms and supports both IPv4 and IPv6. Hosting multiple domains and database redundancy is supported. ser has been carefully engineered with the following design objectives in mind: Speed - With ser, 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. The speed has been achieved by extensive code optimization, use of customized code, ANSI C combined with assembly instructions and leveraging latest SIP improvements. When powered by a dual-CPU Linux PC, ser is able to process thousands of calls per second, capacity needed to serve call signaling demands of Bay Area population. Flexibility - SER allows its users to define its behavior. Administrators may write textual scripts which determine SIP 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. Extensibility - SER's extensibility allows linking of new C code to ser to redefine or extend its logic. The new code can be developed independently on SER 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 SER core compact and fast. Portability. ser has been written in ANSI C. It has been extensively tested on PC/Linux and Sun/Solaris. Ports to BSD and IPAQ/Linux exist. Interoperability. ser is based on the open SIP standard. It has undergone extensive tests with products of other vendors both in iptel.org labs and in the SIP Interoperability Tests (SIPIT). ser powers the public iptel.org site 24 hours a day, 356 days a year serving numerous SIP implementations using this site. Small size. Footprint of the core is 300k, add-on modules take up to 630k.

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.

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. 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.

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. 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.

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. 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. SIP Express Router's support for programmable routing and accounting efficiently allows for implementation of such a scenario.

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). 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. 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.

The following items are not part of current distribution and are planned for next releases: Script processing of multiple branches on forking ser'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. List of known problems is publicly available at the ser webpage at http://www.iptel.org/ser/ . See the "ISSUES" link.

ser is freely available under terms and conditions of the GNU General Public License.

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. Additionally, help may be obtained from our user forum. The community of SER users is subscribed to the serusers@iptel.org mailing list and discusses issues related to SER operation. Public archives and subscription form: http://mail.iptel.org/mailman/listinfo/serusers To post, send an email to serusers@iptel.org If you think you encountered an error, please submit the following information to avoid unnecessary round-trip times: Name and version of your operating system -- you can obtain it by calling uname -a ser distribution: release number and package ser build -- you can obtain it by calling ser -V Your ser configuration file ser logs -- with default settings few logs are printed to syslog facility which typically dumps them to /var/log/messages. To enable detailed logs dumped to stderr, apply the following configuration options: debug=8, log_stderror=yes, fork=no. Captured SIP messages -- you can obtain them using tools such as ngrep or ethereal.

Most up-to-date information including latest and most complete version of this documentation is always available at our website, http://www.iptel.org/ser/. The site includes links to other important information about ser, such as installation guidelines (INSTALL), download links, development pages, programmer's manual, etc. A SIP tutorial (slide set) is available at http://www.iptel.org/sip/ .