SIPWISE CE - NEWS

Sipwise is hiring VoIP engineers (m/f) around Vienna

You will implement efficient and secure call routing features in large-scale systems over geographically distributed components. The job requires very deep knowledge of protocols like SIP, RTP and T.38, a strong experience in writing OpenSER/Kamailio configurations and scripting languages and you should be used to Debian GNU/Linux and SVN/GIT. [ Read More → ]

Sipwise forms Joint Venture with Asipto

In the current era of a rapidly changing telecommunication world, Asipto and Sipwise are pleased to announce the merge of their IP telephony system offerings in order to strengthen the position in the market and to consolidate the development of their unified communication solutions. With Asipto’s know-how and role in the SIP routing development process,…

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sip:providerCE 2.2 – a quick preview

Since the initial release of the sip:provider CE, version 2.1, we’ve mostly worked on two things regarding improvements towards version 2.2: Usability of the Admin Panel, and how NOT to do it: It seems to be a general rule for soft-switch vendors to keep the administrative panel as ugly as possible. There could be different…

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SIP beyond VoIP – location based services using sip:providerCE

SIP is famous for being the protocol to be used for VoIP signaling nowadays. With the (not so simple) SIMPLE extension, it also offers PUB/SUB and messaging capabilities. The plan was obviously to lay grounds for a full-blown communication framework for telephony, instant messaging and presence, as we know it from XMPP. Due to various…

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The sip:providerCE launch – A one-month recap

Back in December 2010 we were launching the sip:providerCE as an open-source offspring of the sip:provider PRO to bring the Sipwise NGCP technology to a much broader audience. During this month, we got a lot of feedback, and it’s time now to look at the impact for Sipwise and the market. How did it go?…

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Erlang for Christmas Shoppers: Why the other checkout line is faster than yours

The Erlang-B calculation is used in telephony systems to describe the probability of call loss on a group of circuits without call buffering. It is however not limited to telephone networks, since it describes a probability in a queuing system. Based on this concept, Bill Hammack from the University of Illinois explains why the other…

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VMware image for sip:provicerCE available

Due to the great success of the VirtualBox image, we’ve created a VMware image of sip:providerCE for quick testing of our free product. Please check out the documentation regarding how to use sip:providerCE virtualization images.

Running sip:providerCE on Virtualbox in Windows

The demand for testing the sip:providerCE on Virtualbox in Windows is actually bigger than expected, and there came up a couple of issues we’re currently working on: The default version when downloading Virtualbox is 3.2.12, which does not support the OVA format (which in fact is just a TAR of the OVF file together with…

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sip:providerCE – the first release of the open-source Class5 VoIP platform

We at Sipwise are excited to announce the first public release of the sip:provider Community Edition (CE). It is a SIP based Class5 VoIP soft-switch, providing every component an operator needs to offer VoIP services. It comes as a communication platform leveraging the capabilities of Kamailio, SEMS and Asterisk, complemented by our own open-sourced building…

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