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
Author: razvanc
OpenSIPS Core Developer
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 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
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
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
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
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
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
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
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
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