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
Author: razvanc
OpenSIPS Core Developer
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
When running a non-interactive service (daemon), such as OpenSIPS, a mandatory requirement is to have a method to interact with it. This is useful for different purposes, such as monitoring the service, provisioning certain data within it, or simply controlling different behaviors of its components. For these matters we have developed the new OpenSIPS CLI tool, an … Continue reading New OpenSIPS CLI tool for the New Management Interface in OpenSIPS 3.0
While the SIP protocol is one of the most popular protocols used for voice calls, the SMPP (Short Message Peer-to-Peer) is one of the most widely used protocols for sending text messages. Having both of them offered by your service enhances your platform with more compatibility and flexibility. In order for your customers to have … Continue reading Gateway between SIP and SMPP messages
The advantages of doing Load Balancing and High Availability without any particular requirements from the clients side are starting to make Anycast IPs more and more appealing in the VoIP world. But are you actually using the best out of it? This article describes how you can use OpenSIPS 2.4 to make the best use of … Continue reading Full Anycast support in OpenSIPS 2.4
The ability to record the calls that go through your platform is gradually shifting from being a feature to being a necessity. Whether you run a call center and you need it to monitor your agents' activity, to comply with your countries laws, or simply to improve your services, you need a recording solution that … Continue reading Call Recording in OpenSIPS 2.4 using SIPREC
