Daniel,
I guess the cat's out of the bag! We've been using OpenSER for quite some time. I'm very happy with it's performance and flexibility. New features are introduced quite quickly without creating a big unstable mess. The performance is very good, we are able to route and account for calls predictably.
One of my favorite things to do around here is to watch the traffic labs stress the call routing implementation. We are able to sustain 60 call per second and burst to 200. It's fun to watch (I know, I'm a geek).
Anyway, we are looking forward to the 1.2 features. Specifically, we will be giving the naptr/srv stuff a run for the money...
Thanks to you, Bogdan and the whole crew for all your hard work!
---greg Greg Fausak lfausak@cisco.com
On 2/22/07, Daniel-Constantin Mierla daniel@voice-system.ro wrote:
Hello,
OpenSER was chosen to be the SIP proxy of Cisco Service Node for Linksys One. That shows the very good level of stability and flexibility as well as wide set of capabilities in OpenSER:
http://www.openser.org/index.php?option=com_content&task=view&id=90&...
I would like to express my thanks to all supporting the project by any means (coding, testing, community support ...). Release 1.2.0 is knocking at the door -- it is a huge step forward since 1.1.0 adding lot of new stuff, plus a handful of improvements to existing features, just to remember some of news in 1.2.0:
- SIMPLE Presence support
- SNMP
- PERL API
- JAVA SIP Servlet Application Server (http://wesip.eu)
- script variables
- DNS failover
- IP blacklists
- XMPP IM gateway
- XMLRPC
- SIP session timers
- secure federation peering
- retransmission timer accuracy
- automatic error handling
So, here I wanted to get: help with testing 1.2.0 is very appreciated these days :-)
For those looking forward to meet other OpenSER folks, the next two major events are:
- Fosdem, Brussels, Belgium, this weekend
- Von Spring, San Jose, USA, in March
Cheers, Daniel
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