San Francisco, Silicon Valley, Trip

I am going to spend few days in San Francisco, during May 06-11, 2011, visiting businesses and friends around.

If you want to meet and chat about Kamailio, SIP Express Router, SIP:Provider Platform or SIP and VoIP in general, drop me an email at

• < miconda [at] gmail.com >.

Two technologies striking back in VoIP - part I

Living a Deja-Vu?!? Or maybe this time the dreams (of IETF/ITU/ETSI specs writers) come true ... who knows. This is a two parts story, about two technologies which many thought went dead long time ago but they are, again, actual in VoIP: IPv6 and IMS (Internet Multimedia Subsystem).

IPv6, the old story is on first pages again. Back in 2002 I worked at FhG Fokus Research Institute in Berlin, developing SIP Express Router (aka SER), within a majority of European Commission funded projects focusing on IPv6.

One may think this is not strictly related to VoIP, well, it is then at least very related to SIP, the leading routing protocol in VoIP these days. Why? Because SIP was actually designed for IPv6 networks. Why? Don't ask me.

Just slightly off-topic note, to refresh, for the past 10 years we had to invent, smash, re-invent solutions and do workarounds to get VoIP going through IPv4 networks -- or more specific saying, through NAT routers, since NAT was not considered in SIP specifications at all. Mixing STUN, TURN, COMEDIA and RTP relays, slipping on ICE, mangling headers and SDP, we got it working. Not few times, friends know that well, it had to be very specific hack per user agent or even firmware.

Back to IPv6, in Kamailio it has been IPv6 support to the core and main modules since May 2002, during the age of SIP Express Router v0.7.y, that's 9 years ago. Looking at the mailing list archive, it is now obvious the interest in the IPv6 is for the first time serious. Of course, the engine behind is the IPv4 address space shortage. Still there is no significant move towards IPv6 as reported recently. There are many (of us) not that happy to remember hex-based IP addresses, which may not be short at all, by the way.

If I hate (phone, digit-based) numbers, preferring instead SIP addresses (username@domain), think about the love I care for hex-numbers.

Moreover, fixing SIP for IPv4 somehow broke SIP for IPv6. Why? Because now we have to support all the nasty tricks specified to make SIP work for IPv4.

The IPv6 was started about a dozen years ago, initial battle was lost, now it is another one. Will it be the final battle? If IPv6 doesn't win this time, then the war is over for it.

To win, it requires fast and smart moves:

• first and most important, is the adoption of IPv6 at the access level - core networks and home routes
• then SIP devices (hard and softphones) with proper IPv6 support
  

If these two are not solved quickly, IPv4 shortage might be workarounded.

The last idea has a strong reason, non-technology related, but at the same time the biggest enemy for IPv6: the cost to switch to it. Internet is way bigger than 10 years ago, replacing/upgrading every piece running IPv4 now is going to be amazing expensive. Time was not the ally for IPv6 at all.

In the next post in this series it will be about IMS, yet another hot topic these days, might not be for VoIP, but surely is for VoLTE...

Maybe in the future we will get to the same situation for Enum? What about mobile IP?!?

Kamailio Booth at LinuxTag 2011

Kamailio - the Open Source SIP Server project has a booth at LinuxTag, Berlin, Germany, May 11-14, 2011.

Location is Hall 7.2b, booth 112 (near to the Mozilla project).

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

• Daniel-Constantin Mierla (Asipto)
• Henning Westerholt (1&1)
• Carsten Bock (ng-voice.com)
• Andreas Granig (Sipwise)
• Stefan Sayer (SEMS)
• Raphael Coeffic (Tekelec)

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

Kamailio v3.1.3 RPMs Available

The RPM packages for Kamailio recently released v3.1.3 are now available for CentOS, Fedora, RedHat and OpenSUSE, available on the opensuse.org build service:

• https://build.opensuse.org/package/show?package=kamailio&project=home%3Akamailio%3Atelephony

Specific download links are:

• http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_CentOS-5/
• http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_RHEL-5/
• http://download.opensuse.org/repositories/home:/kamailio:/telephony/Fedora_14/
• http://download.opensuse.org/repositories/home:/kamailio:/telephony/openSUSE_11.4/

Kamailio v3.1.3 Released

Kamailio SIP Server v3.1.3 stable is out – a minor release including fixes in code and documentation since v3.1.2 – configuration file and database compatibility is preserved.

Kamailio (former OpenSER) 3.1.3 is based on the latest version of GIT branch 3.1, therefore those running 3.1.0, 3.1.1 or 3.1.2 are advised to upgrade. There is no change done to database structure. On 64b operating systems you may need to update the path to loaded modules in configuration file — see the bottom note for more details.

Resources for Kamailio version 3.1.3

Source tarballs are available at:

http://www.kamailio.org/pub/kamailio/3.1.3/src/

Detailed changelog:

http://www.kamailio.org/pub/kamailio/3.1.3/ChangeLog

Download via GIT:

# git clone –depth 1 git://git.sip-router.org/sip-router kamailio
# cd kamailio
# git checkout -b 3.1 origin/3.1
# make FLAVOUR=kamailio cfg

Binaries and packages will be uploaded at:
http://www.kamailio.org/pub/kamailio/3.1.3/

Modules’ documentation:

http://www.kamailio.org/docs/modules/3.1.x/

What is new in 3.1.x release series is summarized in the announcement of v3.1.0:

http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes/

Note: on 64b operating systems, the internal libraries and modules are installed under ‘lib64′ folder (e.g., /usr/local/lib64/kamailio/modules/). If you want to install under ‘lib’ directory, then use ‘LIBDIR=lib’ when compiling and installing (e.g., ‘make FLAVOUR=kamailio LIBDIR=lib install’). If you keep the default lib64, then you may need to adjust ‘mpath’ (or ‘loadpath’) parameter in your existing configuration files and change ‘lib’ to ‘lib64′.

If you want to see what is new in development version (to become the future major release v3.2.0), visit the web page:

• http://sip-router.org/wiki/features/new-in-devel

Kamailio at Google Summer of Code 2011

For the second year, Kamailio project has been selected in Google Summer of Code program, now for year 2011, within Jitsi organization (former SIP Communicator project).

This time the participation is targeting the implementation of a SIP-to-Jingle gateway. Jingle is the voice extension of XMPP. At this moment, Kamailio includes a SIP-to-XMPP gateway for Instant Messaging and Presence, therefore the GSoC project is looking to extend it for voice.

Read more about the project and, if you are a student and interested to implement the project, see how to apply at:

• http://www.kamailio.org/w/gsoc-2011/

SIP Communicator Renamed to Jitsi

Here is a short news just to inform our community that the SIP Communicator SIP softphone was renamed to Jitsi. There are several pages mentioning Kamailio and SIP Communicator out there, among them our participation to Google Summer of Code 2010 as well as the full SIP SIMPLE Presence configuration with XCAP:

• http://asipto.com/u/sp

To fetch the latest version of Jitsi SIP softphone, just go to:

• http://www.jitsi.org

Note: Kamailio has applied for the second year for participation to GSoC, now for the 2011 edition, again together with Jitsi project. Stay tuned for updates on this topic!

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:

• http://ftp.iptel.org/pub/sems/1.4/1.4.0/