OpenSER project is dead: Welcome to Kamailio (and OpenSIPS)

As an open source sip client library, pjsip needs to connect to a server (well, P2P SIP is of course a possibility, especially using NAT Traversal, but that’s a topic for another day).

OpenSER is one such server. A quick unscientific trawl through pjsip mailing list archives reveals more than 80 mentions of OpenSER. We’ve used it ourselves for testing purposes.

For the last week or so I’ve been hearing that OpenSER has been “renamed Kamailio”, according to the news at Kamailio website. That seems to be true, because I can see most of the site, logo, footer is still saying openser.org. And the website openser.org will redirect you to kamailio.org.

So, goodbye OpenSER, welcome Kamailio.

All finished? Not quite.

Welcome also to OpenSIPS (Open SIP Server), which is a “a continuation of the OpenSER project”.

Confused? Don’t be. This sometimes happen in an open source project. The same OpenSER code is taken by both Kamailio and OpenSIPS and from now on will take a life of its own. This is called a ‘fork’. In fact both projects will start with release 1.4.0 (with OpenSIPS releasing today, and Kamailio planning for later this week).

For completeness, I will mention that OpenSER itself is based on SIP Express Router (SER) project.

Is this bad? Depends. If for example you have 100 contributors to OpenSER, and assume it is an even split between OpenSIPS and Kamailio, then you will have ‘only’ 50 contributors each. By simple math it would seem each project is for the worse, because less contributors means less feature implementation, less testing, less everything in general.

But look further then in software projects it is often not the raw numbers that matters. If the 100 contributors rarely agree on anything, then the project is stalled anyway. By forking, then maybe each new segment will be more energised, and will release better software.

What about users? It does leave them in a bewildered state for a while. At the initial release, both projects are almost literally the same. Over time each project should take its own trajectory and we can then evaluate each on their merits.

Where does this leave pjsip as a SIP client framework? Well, as long as all the SIP servers adhere to published RFCs, and have commitment to follow the standards (which they all do), we’re fine. We should be able to connect to SER, OpenSER Kamailio, OpenSIPS, and others that may or may not come after these products/projects.

At the end of day, each open source project, including pjsip itself, lives and die by its adoption.

What do you think? Will you use Kamailio or OpenSIP or none of them? Let us know in the comments.

8 Responses to “OpenSER project is dead: Welcome to Kamailio (and OpenSIPS)”


  1. 1 saghul 5 August 2008 at 0:29

    I’ve been using OpenSER for a while, and now I have to tell I’m a little confused :-/

    I think I’ll first try Kamailio, as it seems to be just a rename, not a fork, and I’m really happy with OpenSER, but I’ll follow OpenSIPS updates very closely, to see how it evolves.

    That’s my 2 cents 🙂

  2. 2 ginel 7 August 2008 at 17:26

    The http://www.openser.org website now points to http://www.opensips.org, so i guess it will be OpenSIPS after all.

  3. 3 lsi 4 September 2008 at 19:31

    is it availible in German?

  4. 4 salman 20 September 2008 at 23:43

    just trying opensips …really confusing..but maybe worth to try..

  5. 5 pedram 24 November 2008 at 12:46

     In an ideal world all equipment would be SIP-RFC compatible which means that when their server changes address it sends a message to the equipment at your end to say where it is moving to. I am very new to working around with pjproject-1.0-rc2 so any help would be appreciated. I need some idea in SIP client/server transactions about how to handle when many SIPusers are approaching a single user(SIP Server). One of the setup requirements is the SIP Proxy IP address (the DNS is given on their site as sip.internetcalls.com)but I am unable to find this detail anywhere.
    I want to be able to use Proxy server capabilities under my tests. Generally, what you’d do is use a full on Proxy server to achieve call between two clients by its, All I need is a SIP Proxy of sorts. A piece of software which will happily connect to my Pjsip in order to recieve incoming calls, and then spit them back out at exchange TCP SIP port when they do arrive.
    Does anyone know of a piece of software which can do this?
    and how we can a callsetup between two clients by proxy server that call flow include RTP packet between two clients and no between proxyserver and client
    Thanks in advance

  6. 6 Symbian Blogger 16 May 2009 at 20:55

    I found your blog on google and read a few of your other posts. I just added you to my Google News Reader. Keep up the good work. Look forward to reading more from you in the future.

  7. 7 spatter 23 June 2009 at 13:16

    The NDSi is the best handheld ever imo, I don’t care what those PSP fanboys say….

  8. 8 Bri@n 21 July 2011 at 13:48

    Hello,
    Maybe you are interested in using an SDK that works with Kamailio:
    http://www.voip-sip-sdk.com/p_22-how-to-setup-ozeki-voip-sip-sdk-with-kamailio-voip.html
    I’ve already tried it, and I’m pretty satisfied with it. Hope you’ll like it!


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