Kamailio (SER) SIP Server v4.2.0 Released!

#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.

SIPExchange 1.1.0b From CafeSIP Is Ready For Download

SIPExchange 1.1.0b snapvoip.blogspot.com
SipExchange is a project by CafeSIP and we have been following the CafeSIP projects for a while. In addition to SipExchange, they also produce;

I is a framework for developing SIP server-side applications using the Java programming language.
SipUnit is a unit testing tool for SIP server-side applications and a bunch of common projects.

SipExchange, a feature-rich SIP softswitch, (a open-source softswitch) that provides standard SIP services like location, proxy and presencethat can be used by service providers to offer voice, video, chat and other multi-media services to their subscribers. SipExchange could be used in many different ways to manage your communication needs;

1. Service providers can use SipExchange to offer SIP-based voice, SIP-based video and data services to their customers.
2. Internet service providers can host specialized SIP services for their corporate customers using SipExchange.
   
3. SipExchange can be used as an IP PBX for communications within an office or even inside a home.
4. Companies can provide specialized SIP-based services to their customers and partners by extending SipExchange.
5. Software vendors can offer specialized SIP-based services to their customers by extending SipExchange.

 You could download SipExchange and other products from CafeSIP.org

OpenSIPS 1.5.0

OpenSIPS 1.5.0 Released
This slipped under our notice but the OpenSIPS team released the latest OpenSIPS server version 1.5.0 on March 23rd, about six month since the last major version release.
While there are are major improvements in Database, Management interface and dialog support, the new additions to the SIP server stands out. These include cache support, Load Balancing, PrePaid support, SIP Identity , Dynamic Routing, IP geo location, to name a few but you can find all the new additions on what is new on 1.5.0.
The team has compiled a manual to help version 1.4 users to migrate to the new OpenSIPS 1.5.0 and is very helpful as there are some major changes need to be addressed such as the new database structure. (but it is much easier than it sounds!)
A set of links, release notes, OpenSIPS Download and other helpfule links are available from the release news page.

Tags: OpenSIPS 1.5.0, SIP Server

OpenSER 1.2.3 Released.

OpenSER has released a new member of the 1.2 family, OpenSER 1.2.3. It brings all kind of fixes (mainly from the 1.3 version) to the 1.2.X branch. Site also noted that no new featues were added. I missed this as I was using the 1.3 from the SVN. Need to visit the site much often. Thanks team now I will inform my clients that use 1.2 branch.

Enhancements and fixes in 1.2.3 includes:

• small fixes (compile warnings, porting across UNIX platform errors)
• bug fixing
• mi_xmlrpc update to work with latest 1.0 libxmlrpc-c3 releases
• functionality fixes (Route header handling in local sequential requests)
• small improvments - more self - sanity checks, more verbous error codes in script (like t_relay() in TM).

All the downloads are available from OpenSER Download.


tag: OpenSER, SIP Server, OpenSER 1.2.3