SIP proxies, by definition, lack the ability to do any media handling, due to the fact that they only stand in the path of call signalling, not call media. Therefore if you want to enhance your SIP proxy with any extra media capabilities, you have to inject a new component in the media path, that … Continue reading Enhanced media capabilities in OpenSIPS 3.1
We are all traversing harsh times, full of uncertainties. Nevertheless, we are doing our best (and yes, we also feel the impact of these times) to keep you all up to date with the progress and planning of the new OpenSIPS 3.1 major release. The Status Starting from the original planning of the 3.1 version, … Continue reading OpenSIPS 3.1 – interim update
Unfortunately, these days are not safe days for gatherings. The pandemic that hunts the whole word makes us think of the health risks. The closed borders, the quarantines make us think of the traveling risks. Too many risks, considering that the purpose of an event such as OpenSIPS Summit 2020 is to bring value and … Continue reading OpenSIPS Summit 2020 – the September replaning
Overview This post is a sequel to the initial write-up on the recently introduced qrouting (Quality-based Routing) module. During the last month, the module has received several key additions, aimed at both improving the data format (gateway statistics, thresholds and scores) as well as the runtime behavior, with a new traffic balancing algorithm having been … Continue reading Quality-based PSTN Routing in OpenSIPS 3.1 LTS [part 2]
DTMF (acronym of dual tone multi frequency) is the signal that your phone sends to your carrier when you press your phone's touch keys during an ongoing call. The primary usage of DTMF tones is to control IVR (Interactive Voice Response) applications, by using the tones to choose certain options from a list. From a … Continue reading Call control using DTMF in OpenSIPS 3.1 LTS
Introduction Note: some information in this post may be outdated. Make sure to also read the follow-up post for the final view on qrouting's behavior. Up until today, the open-source SIP server garden seems to have yielded quite a handful of ways to perform PSTN routing. Some VoIP operators prefer to pull the latest rate … Continue reading Quality-Based PSTN Routing in OpenSIPS 3.1 LTS [part 1]
There is no doubt about the danger and security threats presented by the robocalls or identity thieves. Also there is no doubt that STIR/SHAKEN is the solution that addresses the threats. And the most important aspect of this solution is the fact that this is a standardized, well accepted and (nevertheless) viable solution (see SIPNOC2019). … Continue reading Shaken, not stirred
In order to provide secure SIP communication over TLS connections, OpenSIPS uses the OpenSSL library, probably the most widely used open-source TLS & SSL library across the Internet. The fact that it is so popular and largely used makes it more robust, therefore a great choice to enforce security in a system! That was the … Continue reading The OpenSIPS and OpenSSL journey
As you begin to write or migrate to OpenSIPS 3.x configuration files, you will notice some major changes in the way module functions are invoked. These changes are part of an ongoing effort to improve the experience of writing opensips.cfg code, and in this article we're going to take a closer look at how this … Continue reading The Module Function Interface Rework in OpenSIPS 3.0
Re-homing represents the ability to move a call from one server to another, without causing any disruptions in the endpoints call experience. Although this was already possible using previous versions of OpenSIPS, the setup required to comply with certain network constraints, making it impossible to use in geo-distributed setups. However, starting with OpenSIPS 3.0, you … Continue reading Re-homing your calls with OpenSIPS 3.0
