Showing posts with label yate. Show all posts
Showing posts with label yate. Show all posts

Friday, June 10, 2011

sipgate CEO Explains Why They lected Yate Open-Source VoIP Telephony Engine At VoIP User Conference.

http://snapvoip.blogspot.com/
sipgate CEO Thilo Salmon returns to Voip User Conference (Live, weekly on every Friday 12:00 Noon Eastern Time) this week with Diana Cionoiu, to speak about sipgates use of Yate, an open source VoIP IP Telephony engine from Null Team, Romania.

Following is the press release by SIPGate regarding the YATE.
05/04/2011 - sipgate has chosen Yate, the open source project of NullTeam, to power the core of it's soft-switch infrastructure.
The open-source telephony engine Yate is making inroads into what used to be the domain of proprietary equipment provided by large telecom suppliers. Following the certification of Yate SS7 stack by Deutsche Telekom's test lab, sipgate has started to deploy Yate at the core of it's softswitch. The decision came following extensive tests to determine stability, scalability as well as resilience.
"The open source telephony platform was easily integrated in the infrastructure of our softswitch", said Thilo Salmon, CEO of sipgate. "We are mostly impressed by the quality and scalability of Yate technology."
"We are pleased that sipgate has chosen Yate for implementing their telephony softswitch", said Diana Cionoiu, managing director of NullTeam. "This is a solid recognition of our 7-years development of Yate platform"
About sipgate:
With offices in Austin, Texas and Dusseldorf, Germany, sipgate is a well-established and recognized international telecom company. sipgate has been successfully building its business since 2004. Today, it is arguably the most popular network-independent phone service in a number of European countries catering to a six figure user base.
About Null-Team:
Null Team, based in Romania, is a software company started in 2004, whose founders have years of experience in telephony and software industries. Yate, primary product of the company is a Telephony Engine, usable in a variaty of customized applications. Null Team also sell solutions for call centers, telephony carriers, for enterprise and soho market.

SIPGate, and  YATE via VUC

Monday, March 16, 2009

Load Balance SIP Servers With OpenSIPS

http://snapvoip.blogspot.com/
OpenSIPS could be used to load balance media servers with the "Load Balance" module. The module provides capability to rout traffic based on the load.
The LB module defines the offered resources, and for each resource, it defines the capacity / maximum load as number of concurrent calls the destination can handle for that resource.
For an example derived form OpenSIPS docs;The destinations are assigned with their resources ready to share and as mentioned the capacities. OpenSIPS Load Balancer Module documentation on OpenSIPS Load Balancing - how it works at OpenSIPS provides detailed directions to setup load balancing a set of Yate / Asterisk servers.
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Wednesday, December 17, 2008

The Open Source Ruby Framework Adhearsion

http://snapvoip.blogspot.com/
The open-source Ruby framework Adhearsion (What's Not Telling About Asterisk? ) has updated the roadmap and it is very promising. The day that everyone will be able to make their own VoIP application or platform is coming near. For me the configuration file generators with Asterisk has been the high light. But due to time costraints, testing using has been a few jabs rather than actual productive work. The roadmap.
But the roadmap is entising me to pay more attension to the development process and some more added benefits are coming in the form of FreeSWITCH compatibility and YATE Compatibility.
If you are interested, this tutorial might be of help.