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 → ]
Asipto and Sipwise entered into a joint venture! In the current era of a rapidly changing telecommunication world, Asipto and Sipwise are pleased to announce the merger of their IP telephony system offerings. This is to strengthen the position in the market and to consolidate the development of their unified communication solutions. The combination of…
A sneak peek of Sipwise’s open-source platform: the Sip:providerCE 2.2 Since the initial release of the Sip:provider CE, version 2.1, we’ve mostly worked on two things regarding improvements towards version Sip:providerCE 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…
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…
A month recap after the launched of Sip:provider CE! Back in December 2010, we launched the Sip:provider CE as an open-source offspring of the Sip:provider PRO to bring the Sipwise NGCP technology to a much broader audience. We got a lot of feedback during this month, and it’s time now to look at the impact for Sipwise and…
Erlang: 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…
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…
Sipwise launched the 1st open-source VoIP platform: the Sip:providerCE! We at Sipwise are excited to announce the first public release of the Sip:provider Community Edition or Sip:providerCE. 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,…