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
Tag: SIP
The distributed SIP user location support is one of the major features of the latest stable OpenSIPS release, namely 2.4. The aim of this extension of the OpenSIPS usrloc module is to provide a horizontally scalable solution that is easy to set up and maintain, while remaining flexible enough to cope with varying needs of … Continue reading Clustered SIP User Location: The “Full Sharing” Topology
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
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
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 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
Starting with OpenSIPS 2.2 the registered SIP contacts (stored the location table) have a new unique ID named contact ID. This new ID is contact specific (computed based on various contact elements) and it replaces the old opaque ID which was a simple DB auto-increment key. But why ? to increase the efficiency of the DB operations, by using a single value … Continue reading Migrating registrations to OpenSIPS 2.2
The new mid-registrar functionality is now available with OpenSIPS 2.3 (current development branch) ! What is a mid-registrar ? The mid-registrar is a mid-component of a SIP platform, designed to work between end users and the platform's main registration component. It opens up new possibilities for leveraging existing infrastructure in order to continue to grow (as … Continue reading Mid-registrar: scalable registration and call forking
SIP registration is the process through which SIP user devices (desk phones, soft phones, etc.) periodically announce or refresh their network location to a SIP registrar, for an upcoming period of time. Once registered with the VoIP platform, the device is able to receive calls or messages from other devices. Proxying SIP registrations In some cases, we may want … Continue reading How To Proxy SIP Registrations
Call canceling may look like a trivial mechanism, but it plays an important role in complex scenarios like simultaneous ringing (parallel forking), call pickup, call redirect and many others. So, aside proper routing of CANCEL requests, reporting the right cancelling reason is equally important. How to properly handle CANCEL requests in OpenSIPS? According to RFC 3261, … Continue reading CANCEL request and Reason header

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