Showing posts with label sems. Show all posts
Showing posts with label sems. Show all posts

Friday, May 25, 2012

Kamailio and SEMS in a Risk-of-Life Service

An interesting story sent by Jeremy A. on SEMS mailing list about use of Kamailio and SEMS in emergency services.

This is a belated report on the use of SEMS in a risk of life service.

The system uses Kamailio in a distributed architecture of dozens of Fire & Rescue stations. This is heavily based on distributed and replicated DNS.

A single ’911′ style headquarters has duplicate hot swap-over control rooms at other locations.
The headquarter and alternate posts have servers that service HQ operator positions with SIP phones. These provides sidecar indication of F&R Station state for up to 64 F&R stations – using BLF. These phones are hooked into an integrated analogue audio management system.

Each Fire and Rescue station has an embedded SIP based controller that runs Kamailio and proprietary software to control the F&R station electrical and safety systems as well as provide public address functions to alert the F&R staff of a new emergency. These PA announcements are SIP based using a DSL network and are live from the HQ positions, plus computer synthesized voice, as well as alerting tones.

Each station also has multiple SIP phones for in-station and station to station calling.
The network is decentralized, so failure of the central control system still allows point to point communications between Fire and Rescue stations.

The headquarter systems uses SEMS as the primary operator manager to perform multiple simultaneous deployment calls to remote Fire and Rescue stations. SEMS is used to create a dynamic conference between an operator and multiple Fire & Rescue stations. These are automatically initiated by SEMS and answered by the F&R embedded systems. This means an operator can broadcast a deployment message and initiate station control activities at up to five stations (fifth alarm) This is only constrained by the bandwidth available at the headquarters. Our SEMS packages have been designed to handle non-answered calls to the conference and provide operator indication by ‘SMS’ messages to the handsets and audio feedback.

The system provides full forensic recording by using rtpproxy at all locations. These recordings are archived by an out-of-band process.

Control of the system is purely SIP based – so every item in the system is a SIP based entity. This includes servers, embedded systems, and phones.

The phones are physically integrated into operator positions that also handle PSTN, PBX, and radio traffic. The interface is purely keyboard on the operator phones.
Options for integration of the SIP system into CAD (Computer Aided Dispatch) are obvious. The only drawback is the rusty and ancient systems and the unbelievable process required to get approval to integrate.

The system as provided provides at least 5 nines reliability. Probably a lot better. The only downside is the DSL network (provided by others at amazing expense) that provides a system with a lousy 2 nines reliability. We are in the process of developing an offering using redundant DLS/3G routing to improve this.

The field stations are a hybrid Centos 5/Slax system running out of flash. The HQ systems are straight Centos 5 systems running off disk or off flash. Future versions will be pure Centos out of flash with no fancy memory overlay – flash is well good enough.

The system has been live for over a year with no major issues. I can’t say how many lives have been saved, but certainly quite a few. At least we haven’t been sued yet!

Tuesday, May 8, 2012

SEMS v1.4.3 Released

On the 4th of May, 2012, the SIP Express Media Server project announced the availability of the SEMS 1.4.3 release.

This is a SEMS 1.4 series bugfix release and should be a drop-in replacement to 1.4 installations. For details of the changes see:
Sources are available at:
More about the project:
SEMS is a project tight related to Kamailio, started at the same institute in Berlin, Germany, having many shared developers. It can be used together with Kamailio to provide services such as voicemail, audio conferencing, announcements, IVR menus or back-to-back user agent functionality.

Friday, August 26, 2011

SEMS v1.4.2 Released

The SIP Express Media Server (SEMS) version 1.4.2 is available. Users of SEMS 1.4 versions and anyone else is recommended to upgrade to that version. Please find below the relevant changelog, the source tarball is available at:

Debian/Ubuntu packages:

Changelog for SEMS 1.4.2 release:
* auth’ed BYE (wait_for_bye_transaction)
* fixes SIP auth for qop header format
* xmlrpc: fix busyloop with keep-alive
* a few minor SST issues
* builds on Ubuntu 11.4 (build-deps)
* ivr: release GIL on blocking file I/O
* SBC: fix codec filter for unnamed payloads<96
* fixed DSM variables to outgoing call
* some examples and documentation added

This release introduces a new configuration option wait_for_bye_transaction=yes which enables authentication of BYE requests sent by SEMS, which is disabled by default. If this option is not enabled, SEMS behaves like in 1.4.1.

Tuesday, June 7, 2011

A look at SIP:Provider CE v2.2

SIP:Provider Community Edition (SPCE) has recently released the version 2.2. The out-of-the-box VoIP service operating platform added in this version a lot of interesting new stuff.

First is about the upgrade of the operating system to Debian Squeeze. Also Kamailio SIP Server and SIP Express Media Server (SEMS) are integrated with their latest stable branches.

From this point of view, having the latest Kamailio opens the doors to add by yourself any of its features in version 3.1.x directly in the configuration file, such as SIP/SIMPLE presence or secure communication over TLS.

In terms of architecture, the platform was re-sketched from grounds. It runs an instance of Kamailio to guard the other SIP applications, namely the SIP registrar and proxy (another Kamailio instance), the voicemail server (Asterisk) and the back-to-back user agent (SEMS). Besides the role of entry and exit point in the platform, the first instance of Kamailio acts as a load balancer, meaning, for example, that you can add new SIP proxy/registrar servers as you need.

Talking about security, only the Kamailio load balancer is running on public IP, all the rests can run on an internal one, for example 127.0.0.1, making impossible to be accessed from outside, avoiding DoS attacks on them. The load balancer is not using any SQL database, thus is able to absorb impressive amount of SIP traffic, being easy to deal with any kind of attacks. In addition, all the calls are routed through SEMS for SIP signaling topology hiding, protecting the coordinates of core components and the end points.

Caring about security had high priority in this SPCE release, besides those listed above, there are configurable options to protect against scanning and flooding attacks.

A brand new component of the platform is the ngcp-mediaproxy-ng (some notes about it here) which replaces RTPProxy for NAT traversal. The main benefits are in terms of QoS, ngcp-mediaproxy-ng using a kernel module to relay the media packets. The application has been developed in-house, used for many years in production and now released open source under GPLv3 for SPCE.

The web interface got also some fresh air, in particular the administration portal makes more use of web2.0 technologies, improving the user experience.

I am migrating one of the public VoIP services that run Kamailio to SPCE -- then it would be easy to try & feel it quickly. The plan is to go beyond the standard distribution, very likely will have SIP presence and few more features - the targets to be included in the new version of SPCE.

If you want to give it a try by yourself, choose between the APT repository or one of the provided virtual machines images for VMWare of VirtualBox, see details at:
Stay tuned for more updates!

Wednesday, May 18, 2011

Remarks about LinuxTag 2011

This year I attended the last two days of LinuxTag show in Berlin. Kamailio project had a booth at exhibition hall for the four days, but I arrived late for it from the Silicon Valley trip.

However, it was just in time to meet with most of the people that took care of project's booth. In a rare moment, due to spread across the world, there were 5 of the 11 members of Kamailio management team.
From left to right: myself, Elena-Ramona Modroiu (Asipto), Henning Westerholt (1&1), Raphael Coeffic (Tekelec, developer of SEMS project), Andreas Granig (Sipwise) and Carsten Bock (Telefonica/O2).

Henning had a presentation about Linux at 1&1, including remarks about usage of Kamailio. Carsten did the fast track, two slides presentation of Kamailio project.

The number of visitors was balanced during my two days there, Friday and Saturday, with waves of crowds. As usual, many of the well know open source projects were there, such as famous Linux and BSD distributions, CMS projects such as Drupal or Typo3, Mozilla, a.s.o.

Very interesting for us, we learned that the German Federal Office for Information Security (BSI) sponsors an integration project that aims to provide a secure, scalable and easy to use open source PBX platform when they presented the upcoming "Gemeinschaft" PBX version 4 that will use Kamailio and Freeswitch.

Yet another confirmation that secure communications, including SIP over TLS, will have a relevant boost this year in VoIP.

Inside our booth, we offered demos and presentation of various Kamailio use cases:

Monday, May 2, 2011

SIP:Provider CE v2.2 RC1 is out

SIP:Provider CE, a complete open source VoIP provider servicing platform, released v2.2 rc1. It uses Kamailio v3.1 for core SIP routing, Asterisk for voicemail and SEMS for B2BUA applications. See full release notes at:

This version runs two instances of Kamailio, one acting as load balancer and the second for SIP registrar and proxy services.

Among SIP:Provider CE features: user web management interface, administration web interface and monitoring, call forwarding, click to dial, call blocking, speed dial, postpaid billing engine with individual billing profiles, peering, least cost routing, multi-domain, voicemail, IVR and topology hiding. See more details at:

Tuesday, March 15, 2011

SEMS 1.4.0 Released

SIP Express Media Server (aka SEMS) version 1.4.0 has been released. SEMS is a high performance media and application server for SIP based VoIP networks started at the same research institute with SIP Express Router (SER), thus having same origins as Kamailio as well as sharing several developers. SEMS interoperability with Kamailio is therefore guaranteed, being the first option for a SIP B2BUA whenever is required in a Kamailio deployment.

This release of SEMS features a powerful Session Border Controller (SBC) module. From completely transparent B2BUA to customized URI/From/To, strictly filtered (messages, headers, codecs) with RTP anchoring, Session Timer enforcement, prepaid and call timer, the SBC facilitates interconnect and core routing in a simple and secure way. Thanks to the new multihoming support, SEMS can now be employed at the border of the networks.

This addition also allows to overcome the bottleneck of one NIC – giving the possibility to fully exploit SEMS’ great performance. In the app development area, the DSM language has matured to become a viable candidate also for implementing complex application logic, thanks to language constructs like for, if and functions. SEMS can be downloaded in source from its ftp site at:

Monday, December 13, 2010

SIP:Provider CE

Andreas Graning of Sipwise announced today the release of sip:provider community edition – home page link:

It is a open-source SIP based Class5 VoIP soft-switch leveraging the capabilities of Kamailio, SEMS and Asterisk, combined with custom components in order to provide consistent and easy-to-use provisioning, billing and configuration maintenance.

SEMS 1.3.1 Released

SIP Express Media Server (SEMS) released 1.3.1 with several bug fixes and improvements since 1.3.0, among them:
- fixed CMake build scripts
- fix for architectures w/o atomic built-in functions
- add lost accept_fr_without_totag sample cfg option
- fixed missing CRLF in transfer header

The tarball can be downloaded from:
http://ftp.iptel.org/pub/sems/sems-1.3.1.tar.gz

SEMS is a lightweight media server that can be used along with Kamailio to provide back-to-back user agent functionality, voicemail, IVR, audio conferencing, a.s.o.

Thursday, June 3, 2010

Kamailio Booth at LinuxTag 2010

The project has a booth at LinuxTag, Berlin, Germany, June 9-12, 2010.

Location is Hall 7.2, stand 106, near by StrongSwan project and Snom.

Come and meet some of us, we will be glad to chat about Kamailio, SIP Router, SEMS, SIP and VoIP in general. Depending on the day, you can meet at stand:

We will have a demo running latest version and nice flyers to show the capabilities of Kamailio based platforms.

Monday, April 5, 2010

SEMS 1.2.0 Released

Yesterday, Stefan Sayer announces the release of SEMS 1.2.0 (SIP Express Media Server).

Get the source at:
http://ftp.iptel.org/pub/sems/sems-1.2.0.tar.gz
Some debian packages at:
http://ftp.iptel.org/pub/sems/1.2/1.2.0/packages/
Changelog:
http://ftp.iptel.org/pub/sems/doc/current/changelog.html
Documentation got some new getting started tutorials:
http://ftp.iptel.org/pub/sems/doc/current/index.html

For more information see the home page at: http://iptel.org/sems

Monday, September 14, 2009

SIP Router Development Meeting 2009

Next SIP Router Project Development Meeting coordinates:

Date: Friday, October 2, 2009

Place: Berlin, Germany

The event is co-hosted by FhG Fokus Institute and Technical University Berlin at following address:

FhG Fokus, Room 1008
Kaiserin-Augusta-Allee 31
10589 Berlin
- see venue of the location


Among the goals of the meeting:

  • analyze the development and progress so far
    • Kamailio (OpenSER) and SER integration is 99% completed
    • first major release based on SIP router – Kamailio 3.0 – is due in one month – during October
    • OpenIMSCore extensions ready to use with SIP router core
    • what was good and bad?
  • find solutions for conflicting modules and namings
    • database table structures
    • database table names
    • module names
  • future directions
    • versioning and releasing policies
    • targets for next year
  • organizational aspects
    • infrastructure
    • management
  • social networking and business environment

The participation is free of charge for anybody upon registration via short email at:

registration [at] lists.sip-router.org

this being required for proper dimensioning of meeting room and needed logistics. Registration must be done before September 28, 2009.

Who you can meet there:

Who should consider participation:

  • people willing to get a close feeling about project development
  • people willing to understand how and when SIP Router can be used
  • people willing to meet face to face with the others acting within SIP Router project environment

Agenda:

  • developer slot: presentations from developers
  • community slot: presentations from community and business representatives
  • open discussions
  • social networking event

Hacking day:

  • if there is interest from developers, the days before the public meeting can be organized as hacking session, where you get your hands dirty and code around the project. Do not forget to mention your interest in such event!

You can address general questions about the event via email to:

sr-dev@lists.sip-router.org

More details to follow soon! Stay tuned!