Showing posts with label Kamailio. Show all posts
Showing posts with label Kamailio. Show all posts

Tuesday, October 21, 2014

Kamailio (SER) SIP Server v4.2.0 Released!

Kamailio v 4.2#Kamailio (#SER) #SIP_Server

It is hard to believe that it was 14 years ago that we got to play with SER 0.8.x but certainly the server has come a long long way. Looking at the release notes, this is a major release of Kamailio SIP server, as the development team mentions. OpenHub has attributed 250 man years of development by a massive development team. The commits to main branch alone accounted for 20,000.

Very short list of features are below but follow the link after jump for the full details on Kamailio Wiki;

Ten new modules: evapi, db_mongodb, jsonrpc-s, kazoo, ndb_cassandra, ndb_mongodb, nosip, rtpengine (replacing rtpproxy-ng), tsilo, uuid
TCP-based event API controlled from configuration file
Database connector for MongoDB
Embedded JSON-RPC server
Middle layer connector for Kazoo VoIP platform
no-sql configuration file connector for Cassandra
no-sql configuration file connector for MongoDB
Ability to handle no-SIP messages via configuration file event routes
Control module for RTPEngine application used for NAT traversal or encryption/decryption between Plain RTP and WebRTC-SRTP
Ability to add new branches to active transactions on-the-fly (e.g., as soon as a new REGISTER from callee arrives)
Unique identifiers generator for usage inside configuration file
Pool of generic worker processes
acc module can do asynchronous insert of records to MySQL servers
sqlops module can do asynchronous queries to MySQL servers
async can trigger immediate asynchronous execution of routing blocks
Significant improvements to internal memory manager
Create traffic limiting ‘pipes’ on the fly with pipelimit module
Active calls tracking (dialog) can replicate local data and work with remote profiles
Monitoring (via SIP keepalives) and managing active calls can use dedicated timer processes pool to cope better with heavy traffic
Enhancements to remote registrations to allow reloading records from database; add, enable or disable a record at runtime via rpc commands
Topology hiding can use server IP everywhere
Mechanism to detect retransmissions in early phase of configuration processing without creating the Heavyweight transaction structure
Manage muli-part bodies: set, remove or append parts
Ability to increase and track the difference of CSeq when the server is performing proxy Authentication to next hop for calls
Plenty of new features for IMS extensions

the dedicated wiki page.

Thursday, May 17, 2012

SIP Express Router SER History, So far!

SIP Express Router SER History, So far! http://snapvoip.blogspot.com/

SER was our earliest Open Source SIP experience. It is hard to believe that SER is 10+ years. SER and Free SIP account  Free SIP Account Providers List 2013 were our first iptel.org experience. Since then SER has gone on a long trip and there are two main forks, Kamailio (IPTEL.org and SIP-ROUTER.org project) and OpenSIPS.
We are grateful to all those who are involved and extra 'like' goes to IPTEL for Free SIP account service.
Follow the link after the jump to read more about 10 years of SER.


 10 Years of SER

Friday, February 17, 2012

SIP, The Server SIde. Open Source Telephony Servers For SIP Communications

Open Source Telephony Servers http://snapvoip.blogspot.com/

There are many open source telephony software and server software supporting VoIP IP Telephony. Following are the server side of such software. We have deployed, played around and still using the software listed below.

Tuesday, April 13, 2010

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:

Friday, March 12, 2010

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/

Thursday, October 29, 2009

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

Sunday, September 13, 2009

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.

Thursday, January 22, 2009

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.

Friday, October 17, 2008

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



Thursday, August 07, 2008

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).


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!

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