Getting Started With Diameter In OpenSIPS 3.2

This blog post will briefly talk about how RADIUS and Diameter work, how we managed to incorporate Diameter into OpenSIPS 3.2 and what we have built on top of it so far. Short Protocol Intro The RADIUS spec first appeared in 1997 and was revised in 2000. Its purpose is to offer Authentication (Are the … Continue reading Getting Started With Diameter 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

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

Achieving service redundancy in two steps with unified clustering in OpenSIPS 3.0

A hot backup means redundancy, redundancy means more uptime, more uptime means a better SLA, a better SLA means happier customers and more money. Building redundancy is a must when moving your service into production. And a typical approach for achieving redundancy is by implementing an active - backup setup with full realtime synchronization between … Continue reading Achieving service redundancy in two steps with unified clustering in OpenSIPS 3.0

Gateway between SIP and SMPP messages

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

SIP bridging over multiple interfaces

There are scenarios where you need OpenSIPS to route SIP traffic across more than one IP interface. Such a typical scenario is where OpenSIPS is required to perform bridging. The bridging may be between different IP networks (like public versus private, IPv4 versus IPv6) or between different transport protocols for SIP (like UDP versus TCP … Continue reading SIP bridging over multiple interfaces

Handling SIP Redirect Requests in realtime

The SIP redirect mechanism is a simple and straight forward one  - the originally contacted destination indicates, via a 3xx reply, that a different set of destinations should be contacted. The SIP redirect is mainly used for calls (for INVITE requests), even if the RFC3261 does not limit it to that. Usage cases The primary … Continue reading Handling SIP Redirect Requests in realtime

Troubleshooting missing ACK in SIP

We all experienced calls getting self disconnected after 5-10 seconds - usually disconnected by the callee side via a BYE request - but a BYE which was not triggered by the party behind the phone, but by the SIP stack/layer itself. This is one of the most common issues we get in SIP and one … Continue reading Troubleshooting missing ACK in SIP

SIP-I and SIP with OpenSIPS 2.3

SIP-I considerations The SS7 interconnections are always painful. Both as cost and technical difficulty/complexity. So, as a more accessible alternative, the carriers started to offer SIP interconnection via SIP-I (SIP Infrastructure). SIP-I  or SIP Infrastructure (define by ITU) is very similar to SIP-T or SIP for Telephones (defined by IEFT). SIP-I is very powerful when … Continue reading SIP-I and SIP with OpenSIPS 2.3