Running MI commands from script in OpenSIPS 3.2

During this year's annually OpenSIPS Feature Survey, the poll results for the new OpenSIPS 3.2 reflected an emergent need for people to be able to run MI alike commands directly from the script. Initially, we were a bit reluctant from developing this feature due to the fact that (historically speaking), OpenSIPS's Management Interface has been … Continue reading Running MI commands from script in OpenSIPS 3.2

Media high availability/re-anchoring using OpenSIPS 3.2

Using a media relay server (such as RTPProxy, RTPEngine or MediaProxy) in your VoIP system is a rather common requirement due to various reasons: NATted clients media handling, legal compliance (recording) requirements or for offering enhanced services, etc. Thus, in order to provide high availability for your services, you also need to consider it for … Continue reading Media high availability/re-anchoring using OpenSIPS 3.2

Monitoring OpenSIPS using Prometheus and Grafana

Monitoring real time statistics is a great tool to assess the performance of your services, as well as for detecting, and possibly preventing, unfortunate events. And visualizing the monitored statistics in a graph or chart can definitely improve your DevOps team experience, as well as reduce the troubleshooting time of possible failure events. In a … Continue reading Monitoring OpenSIPS using Prometheus and Grafana

Improved series-based call statistics using OpenSIPS 3.2

Real-time call statistics is an excellent tool to evaluate the quality and performance of your telephony platform, that is why it is very important to expose as many statistics as possible, accumulated over different periods of time. OpenSIPS provides an easy to use interface that exposes simple primitives for creating, updating, and displaying various statistics, … Continue reading Improved series-based call statistics using OpenSIPS 3.2

Calls management using the new Call API tool

The new Call API project consists of a standalone server able to serve a set of API commands that can be used to control SIP calls (such as start a new call, put a call on hold, transfer it to a different destination, etc.). In order to provide high performance throughput, the server has been … Continue reading Calls management using the new Call API tool

Re-homing your calls with 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

No Downtime for OpenSIPS 3.0 restarts

Doing maintenance on live production servers is never desired, but it is always necessary in order to enhance your platform with new features, or just for doing some small/quick fixes. And maintenance usually requires restarting OpenSIPS, so that it can re-parse its configuration script. Depending on your setup, OpenSIPS may need, for routing purposes, a … Continue reading No Downtime for OpenSIPS 3.0 restarts