OpenSIPS Certified Professional (OCP) Program Launched.

OpenSIPS (OpenSER) http://snapvoip.blogspot.com/
If You are serving your clients solutions based on OpenSIPS (formarly OpenSER), now you can put them at ease with the OpenSIPS Certified Professional (OCP). The OCP credentials will show that you can design, implement and administer a SIP solution based on the OpenSIPS SIP Server.

Once certified, the credential will cover the abilities of the professional;

knowledge of Installation, SIP Routing, OpenSIPS Scripting Skills, Authentication, OpenSIPS Control Panel, Connection to the PSTN, Call Forwarding and Media Server Integration, NAT Traversal, SIP Server Load Balancing, Accounting and Monitoring Tools. 

To earn the OCP on OpenSIPS Server, you must pass a single exam:

• OpenSIPS Certified Professional Exam

Exam Details:

The OpenSIPS exam is a 60 minute test with 70 questions.

Register for an exam via exams@opensips.org .

Further information is available at OpenSIPS certification information page.

IRC Meeting For Kamailio (OpenSER) and SIP Router Developers On April 14th 2010

http://snapvoip.blogspot.com/
IRC Meeting For Kamailio (OpenSER) and SIP Router Developers will take place (tomorrow) Wednesday, April 14, 15:00UTC, 2010.
The channel is: #sip-router on server irc.freenode.net.
A lot of  development team members are expected to be present. Everyone is welcome to join and participate in the discussions.
A draft agenda is available at:

• http://sip-router.org/wiki/devel/irc-meetings/next

The SIP Router Project, User Conference

The SIP Router Project snapvoip.blogspot.com

 VoIP Users Conference March 19, 2010, 17:00GMT

ZipDX using g722 or g711 sip:200901@login.zipdx.com
ZipDX PSTN: +1 567 252 2286 iNum: +883 5100 123 94882 

The SRP (SIP ROUTER PROJECT) a progressive project born out of Kamailio (former OpenSER) and SER (SIP Express Router) is holding a VoIP user conference, an audio conference which you could attend via SIP, PSTN, skype and other options.
The conference will focus on:

• the achievements of SIP Router Project so far
• the role of Kamailio (OpenSER) release within the SIP Router project
• new features in Kamailio (OpenSER) 3.0
• what it takes to upgrade from older Kamailio (OpenSER) versions
• new features in development version and future plans for SIP Router Project

The team also released Kamailio v3.0.1 recently.
 
http://www.voipusersconference.org/about/future-topics/

http://www.voipusersconference.org/how-to-participate/

VoIP Security, Asterisk, SIP, Brute Force Attacks, Explained By John Todd Of Digium.

VoIP Security snapvoip.blogspot.com
My thoughts when I read the article saying  “Asterisk attacks are endemic” were ????*&^ and I also knew that John Todd knew what he is doing. So I let the article pass as I was certain it was not the case.
Today Todd has posted an article explaining the misinterpretation of his thoughts and the verbalization of the same.

My comment in the article was not that “Asterisk attacks are endemic”, but that SIP-based brute force attacks are endemic.  Every SIP system that is open to the “public” Internet is seeing large numbers of brute-force attacks.  Sites that have weak username and weak password control will be compromised – this is little different than email accounts being taken over by password-guessing systems and used for sending floods of email.  The significant difference is that when someone takes over a SIP platform to make outbound calls, there is usually a direct monetary cost, which gets people’s attention very quickly.

We all know that Asterisk is used world over and is the favorite SIP based telephony platform is likely to attract people who are interested in hacking or attacking the system. But we also have seen security measures taken by Asterisk as well. Just like the one released day before yesterday, AST-2009-07, where the advisory and the fix was released simultaneously.
But what Todd wrote makes sense and as Asterisk, Broadsoft, Cisco, Kamailio, OpenSER, FreeSwitch, Avaya are all vulnerable to brute force attacks and every bit of  information will help. Whether it is news worthy or not.
Asterisk and SIP Security Redux

SIP Router Org, ASIPTO and SIP Router Trining

SIP-ROUTER.ORG

I am glad that the original SIP Router, (SIP Express Router, SER from IPTEL org, where I got my first free SIP Account) is in the limelight again with the corporation with the Kamailio (OpenSER) the original fork from the SER project in 2005. OpenSER project separated into two a while ago into Kamailio and OpenSIPS. I have been using testing both while being l)oyal to SER, where I learned to sip SIP in 2001, where no other open source SIP servers existed (If they did, I did not know.)

 

The SIP Router Project will bring together a common collaboration framework for any project originating from SIP Express Router (aka SER), referred further as x-SER projects. The projects that joined the initiative so far are: SIP Express Router (SER) – the initial project started in 2002 and Kamailio (OpenSER).

In any case I am looking forwared to the SER 3.0 and Kamailio 3.0 which is slated to be released soon. Good work and cheers to both the teams.

Another bit of news that I would like to bring is that ASIPTO will conduct a SIP router training known as SIP Router Masterclass in November 9-13, 2009 in Berlin, Germany.
The teachers are non other than,

Daniel-Constantin Mierla – co-founder of OpenSER/Kamailio project in 2005, currently core-developer and member of project’s management board

Olle Johansson – Asterisk developer and member of the Digium Asterisk Advisory Board.

 From a post on ASIPTO;

The course targets system administrators and people that act in large VoIP/telephony network services, as well as integrators of VoIP, instant messaging and presence with web 2.0 or similar technologies.

Learning to configure the SIP server is not easy, but is the key for a successful and secure  VoIP business. The flexibility of SIP routing engine allows you to implement in no time innovative services, IP telephony, Instant Messaging, Presence and beyond. Asterisk comes to complete with rich media services and applications. Doing everything designed right and scalable saves time and money.

We create the opportunity for you, guided by experienced instructors, to learn how to build an Unified Communication platform from scratch using the SIP server engine and Asterisk.

Link to course details and registration.

Integrating Asterisk And OpenSIPS In Realtime : Tutorial

Integrating Asterisk And OpenSIPS

We have been working with Asterisk and OpenSIPS for a long time (Even when OpenSIPS was OpenSER!). But we had to hammer out solutions as they fit and there were no methodology to it. Various engineers had their notes and were collected at one place only to be pulled out when the need arises to bring these two Open Source Telephony servers together.
SIP server like OpenSIPS usually provides registration and call handling, either by routing, forwarding or direct connections. The Asterisk on the other hand is a full blown telephony server, providing IVR, Voicemail, conferencing and announcements in addition to call handling, So by bringing these two servers (services) together, one will be able to provide full blown telephony solution integrating SIP, and Asterisk services.
So without saying that our method(s) are messy, I am going to point you to a very nicely written (Also very nicely thought out) article on how to integrated Asterisk and OpenSIPS.
Bogdan Andrei Iancu, a core OpenSIPS developer has written the tutorial. Go read it, Print it and save it! It is very good!
PS The tutorial currently addresses Asterisk 1.4 or 1.6 and OpenSIPS 1.5x. Once the OpenSIPS 1.6 is released the documentation will be updated appropriately.
Realtime OpenSIPS - Asterisk Integration

SIP Routing, A Webinar By OpensSIPS

Routing in SIP
We here are big fans of OpenSIPS (Formally OpenSER). In fact it is our favorite SIP engine. We have been involved with this SIP Server ever since it forked from SER.
The OpenSIPS is very capable SIP engine and is capable of handling large volumes of users and traffic. But we tend to make mistakes due to lack of understanding and the OpenSIPS have steped in to help us all. The second in series of webinars will be presented this month.
(If you need further information on OpenSIPS, you can visit OpenSIPS website and There is also a Recorded webinar, Introduction to OpenSIPS. Links all at the end).

Next Webinar;

Routing In SIP

Date: Monday, July 8, 2009
Time: 8:00 PM - 9:00 PM EEST (5:00 PM - 06:00 PM GMT)

This webinar will help you to understand SIP routing concepts. SIP routing is very often a misunderstood concept. It is essential for anyone trying to script OpenSIPS to fully understand these concepts.

Registration form : https://www2.gotomeeting.com/register/429289130

Registratio is free but the seating is limited. So register early.

Former webinar held on June 8th June, Introduction to OpenSIPS, and a video recording of this webinar will help you to get in tune..

Tags: OpenSER, OpenSIPS, SIP Routing, SIP Education, SER, Routing In SIP, Introduction to OpenSIPS

SIREMIS Project, Web Management Interface For Kamailio (OpenSER).

http://snapvoip.blogspot.com/
Kamailio has announced SIREMIS - new web management interface for Kamailio (OpenSER) and SIP-Router.org. The project is headed by up to this point by Apisto, utilizing OpenBiz PHP framework. The idea behind the plan was to make it easier to involve developers from all walks. Following the open source nature, the project sources will be hosted on sourceforge.net and developers will have access to SVN code repository. (project name: siremis). Currently the project is hosted at Asipto.
I have not used the product but have seen some screenshots which to say the least, very promising.
The beta release include following features:
- subscriber, database aliases and speed dial management
- location table view
- dispatcher, prefix-domain translation and least cost routing (lcr) management
- access control lists (user groups) and permissions management
- accounting records, missed calls and call data records view
- hash table, dial plan table and user preferences table management
- off-line message storage view
- it was tested with MySQL but should support other sql-like DBs
Release note.

Tags: SIREMIS, Kamailio, Apisto, OpenBiz PHP, openser dialplan, SIP-Router.org

OpenSER 1.3 and MediaProxy 1.0 Has Reached End Of Life

http://snapvoip.blogspot.com/

Multimedia Service Platform based on OpenSER 1.3 and MediaProxy 1 has reached its end of life. The new platform is based on OpenSIPS 1.4 and MediaProxy 2.0. Customers are required to schedule a platform upgrade to migrate to the latest software versions. The migration requires changes in the SOAP/XML provisioning client. Customers can update their SOAP/XML clients by using AG Projects testing facility at:

https://secure.dns-hosting.info/ngnpro/wsdl

The changelog for the latest software versions is found at:

https://secure.dns-hosting.info/debian/changelogs.txt

Tags: OpenSER, OpenSIPS, MediaProxy, OpenSER 1.3, MediaProxy 1.0, OpenSIPS 1.4, MediaProxy 2.0

Enhanced LCR Module For Kamailio

http://snapvoip.blogspot.com/
Kamailio, the ex OpenSER team has released an enhanced version of lcr module, in to the development trunk.
The new high-performance implementation keeps lcr information in an in-memory hash table. The size of which could be managed with the new parameter 'lcr_hash_size'.
The 'weight' field in 'gw' table allows to assign a gateways according to weight in a group. Among other enhancements, lcr_reload function is now executed under a lock minimizing race conditions.

You can learn more about what is new in upcoming Kamailio 1.5.0 at Kamailio Dokuwiki site:

If you are a developer, then download trunk version of Kamailio via SVN:

svn co https://openser.svn.sourceforge.net/svnroot/openser/trunk kamailio

Tags: Kamailio lcr module, Kamailio, OpenSER, Kamailio 1.5.0

OpenSIPS 1.5.0 Gets A New Routing Module.

http://snapvoip.blogspot.com/
OpenSIPS announced that OpenSIPS 1.5.0 (development branch) has a new module for routing. Even though it is similar to LCR module in basic functionality like routing based on rules, the major difference seems to be in ability to scale and high performance.

The OpenSIPS site noted that the new module is following capabilities and enhancements;

Internal data organized as a search tree which is very fast
Unlimited number of rules and gateways (Tested on a production setup with 200K rules and with no search penalties)
A rule may point to set of GWs to be used for;
(1) sequential selection, (2) random selection , (3) weight selection
Inbound processing - prefix and strip for calls originated by the GWs

The module already sports an intuitive web interface for provisioning which will released in the near future, along with an OpenSIPs Control Panel Web interface.

A comprehensive documentation and description available from the OpenSIPS development site.

Tags: OpenSIPS, OpenSER

OpenSIPS At SIPIT and My Loyalty!

Solarion seems to be head over heals with OpenSIPS but after digging deep into the story behind OpenSIPS, I tend to siding with Solarion. Like Solarion, I also got my SIP feet wet with SER. Even though stiff and don't feel like at home was all around IPTel, We hung around because there were not many other resources providing SIP down to nuts and bolts SER did.
So When OpenSER came out with new development and management philisophy, I was over the wall! I started converting all my SER servers (quite a few) in to OpenSER servers. Even while doing so I could not belive the version numbers climbing up. I was so used to SER being stuck on version 0.7 I think, I had a party when OpenSER reached version 1.0. Because I knew then SER was history stuck is version 0.9.
Now the tables have turned and for what ever the politics or other reasons, OpenSER has been forked and two new entities, OpenSIPS and Kamailio (I feel good now as I thought I was bad at picking names!).
If you have noticed, I still have a link to SER, on the links on the right. I guess it will be there for giving me first real taste of SIP. But otherwise, I have moved on. And like the Solarion, I am leaning towards OpenSIPS and I will be playing with OpenSIPS 1.4 this weekend. May be Kamailio afterwards.

But I know where I will see OpenSIPS. as they have promised to be at SIPIT, so will I. I will see my Mom and then drive to Lannion as I have not done this great past time I learned in the USA, Driving long distances! Yes I know it is expensive and environmentally incorrect, but like Mr. Gore, I pay for my Carbon foot print else where in the world, I have done so even befor being green became the buzz word of the century.

So I picked up the following information from OpenSIPS site;

OpenSIPS at SIPIT

OpenSIPS project is registered to participate the 23th edition of SIPIT - SIP Interoperability Test event in Lannion, France, October 13-17, 2008.
Compliance and interoperability are essential parts of project testing and participating to such large testing events is a natural step in the QA process.

If there are people interesting in being part of the OpenSIPS test team, please let us know (via the users mailing list).

Tags: OpenSER, OpenSIPS, Kamailio, SIPIT, Lannion, Going Green, SIP Interoperability Test

KAMAILIO v1.4.0 Released

Kamailio Server Version 1.4.0 has been released by the Kamailio (formally OpenSER) with new features since OpenSER v1.3.0 as well as improvements to existing components. Much effort was put in code cleanup and stability and new features in core and old modules were added while other 6 new modules were introduced.
You will find the release here.
This gives me a problem now in writing. What used to be SER vs OpenSER, now becomes SER vs OpenSIPS vs Kamailio!

Tags: SER vs OpenSER, SER vs OpenSIPS vs Kamailio, SIP Server

OpenSIPS 1.4.0 is released.

Update: OpenSIPS will be present at ClueCOn Telephony Conference. So lookout for them if you are at the conference. I will.

As you may have already read, I just wrote about OpenSER chnaging the name and the New fork of OpernSER, OpenSIPS. Well you don't have to wait a lot to see the OpenSIPS! They have just released the OpenSIPS 1.4.
The OpenSER/Kamailio was planing to release their what ever 1.4 version of august 7th. But I guess the OpenSIPS were faster. I think I will try it on one of my virtual machines this weekend. According to OpenSIPS site, the release incorporates new functionalities like

• dialog profiling extension
• local route for internally generated requests
• dial plan / translation module
• general NAT traversal module (for non-invite signalling)
• peering and rate limit (traffic shaping) modules

, enhancements like:

• SRV load balancing (based on weights)
• nonce re-usage verification (safer authentication)

and bug fixing - code, specs and functionality bugs.

The full Changelog is available here.

To get the OpenSIPS 1.4.0 version, see the Downloads page.

Tags: OpenSIPS 1.4.0, OpenSER, Kamailio, OpenSIPS, ClueCon

OpenSER Changes Name to Kamailio and gives Birth To OpenSIPS!

I Just got a book on OpenSER from RavenII for reading and to write a review. I have been a fan of OpenSER since it began. Actually I was all over Iptel's SER befor that and when all the developers, well almost all, left to start OpenSER, I too followed.
Since then we have collectively deployed OpenSER in various in Public and private installations. Mostly with combination of Asterisk and Asterisk hardware from Digium (Asterisk being the gateway for OpenSER). In the open source SIP world, besides SER, and OpenSER, there are hardly any other that is useful to us. Yes we have used a few other stacks but non in the level on OpenSER.
Now if you remember, SER (I am still loyal to SER for starting SIP Revolution) was forked to create OpenSER and since OpenSER became Kamailio (how do you pronounce that? Just imagine my plight! Hello customer I am here to service OpenS.. err Kamailio! What? you are doing what??) I think I might stick with OpenSIPS! (if it is as good or better than OpenSER) at least my customers will not hear the difference, OpenSER <-> OpenSIPS OpenSIPS is a fork of OpenSER/Kamailio. If you have not read, don't forget to read OpenSer VS SER!
Now the book, which was published by Packt, OpenSER Book will be reviewed here, unless the publishers change the name!

Tags: OpenSER, SER, OpenSIPS, Kamilio, Asterisk, Packt Publishing