The module implements authentication over JSON Web Tokens. In some cases ( ie. WebRTC ) the user authenticates on another layer ( other than SIP ), so it makes no sense to double-authenticate it on the SIP layer. Thus, the SIP client will simply present the JWT auth token it received from the server, and pass it on to OpenSIPS which will use that for authentication purposes. For more, see the module documentation.

But the full list of goodies offered by OpenSIPS 3.1 (and a more technical one too), together with migration instructions, can be found on the OpenSIPS 3.1 release notes page.

Starting with version 3.1, the Call Center got several improvements to make easier the integration of the module with different external tools for the agent side, to overall improve the agent capabilities and experience in an OpenSIPS based solution. Extensions as call dissuading, enhanced wrapup time, pre-call agent announcement and agent triggered events are just couple of the 3.1 additions.

The SIP Push Notification support was improved and aligned to the requirements as per newly adopted RFC 8599. In the same time, several particularities were addressed (mainly regarding the contact registration) such as privacy concerns or device identification concerns.

This module adds support for implementing STIR/SHAKEN (RFC 8224, RFC 8588) Authentication and Verification services in OpenSIPS. A more comprehensive description is available on the OpenSIPS blog, while the technical details are available in the module's readme.

This is a new module which keeps track of the signaling quality of each outbound gateway at runtime and dynamically alters the gateway ordering in order to ensure an optimal quality for your platform's PSTN termination signaling! (documentation) (blog)

The rate_cacher module provides a means of caching and real-time querying of the ratesheets assigned to your clients and / or vendors. It also allows for real-time cost-based routing and cost-based filtering. What the module is able to do, how to provision and use it, all are nicely described in this blog post.

JWT Authentication

The module implements authentication over JSON Web Tokens. In some cases ( ie. WebRTC ) the user authenticates on another layer ( other than SIP ), so it makes no sense to double-authenticate it on the SIP layer. Thus, the SIP client will simply present the JWT auth token it received from the server, and pass it on to OpenSIPS which will use that for authentication purposes. For more, see the module documentation.

In order to allow a better grip and control over the calls in OpenSIPS, the 'dialog' module comes with two important enhancements. First is about the inter-dialog data exchange, to be able to exchange data between various ongoing dialog. The second is about providing new dialog triggers, to allow a better call monitoring and control directly from the script level.

The "Call API" is a new standalone component added to the OpenSIPS ecosystem to offer an API for controlling the calls going via OpenSIPS. By using an WebSocket based protocol, this GO based API is able to start, terminate, mute/unmute and transfer the calls hosted on OpenSIPS. Note that this API is not to be used as a SIP-end point, but as a call controller. The offered API is bidirectional - while the API accepts the call control commands, the API is also feeding back with events about the call progress or status.

The B2B engine is on the verge of some dramatic long term changes and 3.1 includes the first, very important ones. First of all it is critical to have clustering/replication support in order to address High-Availability or distribution scenarios. Another change was the implementation of the "b2b context", a new mechanism to allow, from the script level, the attaching of custom data to a B2B session. This helps in correlating all the entities (UAC, UAS) part of a B2B session, to share and exchange such data between the entities.

In Class 5 services, BLF is an important feature, so there was a major rework of the module in OpenSIPS 3.1. The target was to 'properly support the call forking scenarios (like call redirects or call hunts), to correctly report all the involved parties in the call and how they (may) change during the call setup. Now the 'pua_dialoginfo' module is able now to get access to the transaction information (not only to the dialog information as before) in order to get accurate information about the call setup.

Starting with version 3.1, the Call Center got several improvements to make easier the integration of the module with different external tools for the agent side, to overall improve the agent capabilities and experience in an OpenSIPS based solution. Extensions as call dissuading, enhanced wrapup time, pre-call agent announcement and agent triggered events' are just couple of the 3.1 additions.

The module enables the handling of the "as-feature-event" event package (as defined by Broadsoft's Device Feature Key Synchronization protocol) by the presence module. This can be used to synchronize the status of features such as Do Not Disturb and different forwarding types between a SIP phone and a SIP server.

DTMF handling

For both RTProxy and RTPEngine, OpenSIPS is able to report to the script level the DTMF events sampled from the passing RTP. This make possible the implementation of simple IVRs and/or authentication via DTMF with nothing more than OpenSIPS and the media relay. Combined with other functionalities such as media exchange or Back-2-Back, you can even do some cool DTMF driven calls scenarios.

In order to allow a better grip and control over the calls in OpenSIPS, the 'dialog' module comes with two important enhancements. First is about the inter-dialog data exchange], to be able to exchange data between various ongoing dialog. The second is about providing [[https://blog.opensips.org/2020/05/26/dialog-triggers-or-how-to-control-the-calls-from-script/|new dialog triggers, to allow a better call monitoring and control directly from the script level.

The "Call API" is a completely new addition to the OpenSIPS ecosystem, it is a new component that offers an API for controlling the calls going via OpenSIPS. By using an WebSocket based protocol, this GO based API is able to start, terminate, mute/unmute and transfer the calls hosted on OpenSIPS. Note that this API is not to be used as a SIP-end point, but as a call controller. The offered API is bidirectional - while the API accepts the call control commands, the API is also feeding back with events about the call progress or status.

This is a new module that provides means to manipulate the RTP streams of the ongoing proxy'ed calls in OpenSIPS. The module is able to inject or to copy the RTP of such proxy'ed calls/dialogs by exchanging its SDP bodies (and RTP streams) with some new UAC calls created (by the module) towards a media server. Using this new module one can inject/playback announcements or music on hold during an ongoing call, or listening a conversation of two different participants.

https://blogopensips.files.wordpress.com/2019/12/opensips-3.1-crafting.jpg Routing calls and handling large volume of traffic is not a challenge anymore for OpenSIPS. The 3.1 release focused on complex call crafting, basically on increasing OpenSIPS's ability to create and handle complex calling scenarios where multiple SIP calls are mixed and able to interact. Or shortly said, the 3.1 release addresses the Class 5 specific calling features and how to control such calling features via APIs.

Call Handling

Call API

Media Exchange

DTMF

Dialog enhancements

DFKS support

Back-2-Back enhancements

BLF extended support

Call Center enhancements

Call Enhancements

Push Notification support

STIR & SHAKEN

Quality Based Routing

Call Rating support

JWT Authentication

Another new OpenSIPS module, that is able to manipulate calls/dialogs going through OpenSIPS, along with UAC/UAS dialogs (call originated or terminated into OpenSIPS), with the sole propose of mixing the RTP media between these multiple flows. The idea is to make possible the injection of streams of media from / to proxied calls, with the help of auxiliary calls initiated by OpenSIPS. Typical example is to play media within an existing call/dialog with nothing more than simple re-INVITEs - OpenSIPS will initial a new sip call to a media server, in order to get the RTP stream for the playback and it will push it into the proxied call by triggering in-dialog re-INVITEs.

As we already have for transactions, the dialog module will allow the script writer to set, in a per-dialog fashion, script routes to be triggered by various dialog events. Similar to t_on_failure(route), you can do dlg_on_timeout(route) to have a route called when the dialog lifetime is exceed. In the route, you may decide to extend the lifetime, to terminate the call or do any other logging. We can foresen dlg_on_answer(route), dlg_on_terminate(route)' (and more) triggers, which will give a better interaction and control over the ongoing calls.

For both RTProxy and RTPEngine, OpenSIPS will be able to report to the OpenSIPS script the DTMF events sampled from the passing RTP. This will make possible the implementation of simple IVRs and/or authentication via DTMF with nothing more than OpenSIPS and the media relay.

In Class 5 services, BLF is an important feature. Besides working out clustering support for BLF, an important task is reworking the BLF implementation to be call-branch aware, to be able to properly report the call events in parallel calling or call hunting scenarios.

Instead of using the XML scenario to drive the B2B logic (mixing between the calls), we want to use the OpenSIPS scripting for this purpose. This will eliminate all the limitations of the XML language (logic and action) and it will tremendously increase the level of integration of the B2B engine with the rest of the OpenSIPS functionalities. Shortly, more complex B2B logic will be possible, and also better integrated with the rest of OpenSIPS.

An important improvement of the B2B engine will be the addition of the B2B context, to help in correlating all the entities (UAC, UAS) part of a B2B session, to attach custom data to the entities and to exchange such data between the entities. This will help with Accounting and media relaying support for B2B, but also with building custom data sharing inside a B2B session (or between the sessions).

• distributed call-center or a geo-distributed single call queue which gets calls via different OpenSIPS instances and which distributed agents connected to different OpenSIPS instances.

Dealing with Robocalling and CLI spoofing becomes a must when building advanced calling solutions. The support for STIR/SHAKEN will be part of OpenSIPS 3.1, providing multiple usage models, in terms of how the certificates are to be handled during the verification process. Also a flexible approach (to the standards) will be able to cope with all the potential changes derived from the adoption process (of the standard by the providers).

Without actually using a Back2Back model, but simply operating with calls (dialogs, UACs or UASs), many complex class 5 calling (and call mixing) scenarios may be scripted.

A new OpenSIPS module, placed on top of dialog module, will allow remote control over the calls going through OpenSIPS. The module will expose a simplified set of commands (API like) for setting up new calls, for answering and terminating calls, for transferring or putting on-hold calls - all these without interacting with the end-devices, but triggering and handling the action only from the OpenSIPS (as proxy) level.

Another new OpenSIPS module, that is able to manipulate calls/dialogs going through OpenSIPS, along with UAC/UAS dialogs (call originated or terminated into OpenSIPS), with the sole propose of mixing the RTP media between these multiple flows. The idea is to make possible the injection or stream of media from / to proxied calls, with the help of auxiliary calls initiated by OpenSIPS. Typical example is to play media within an existing call/dialog with nothing more than simple re-INVITEs - OpenSIPS will initial a new sip call to a media server, in order to get the RTP stream for the playback and it will push it into the proxied call by triggering in-dialog re-INVITEs.

B2B engine

Improve the media

Routing calls and handling large volume of traffic is not a challenge anymore for OpenSIPS. The upcoming 3.1 release will focus on complex call crafting, basically on increasing OpenSIPS's ability to create and handle complex calling scenarios where multiple SIP calls are mixed and able to interact. Or shortly said, the 3.1 release will address the Class 5 specific calling features and how to control such calling features via APIs.

Dealing with Robocalling and CLI spoofing becomes a must when building advanced calling solutions. The support for STIR/SHAKEN will be part of OpenSIPS 3.1, providing multiple usage models, in therms of how the certificates are to be handled during the verification process. Also a flexible approach (to the standards) will be able to cope with all the potential changes derived from the adoption process (of the standard by the providers).

Push Notification (RFC8599)

The existing PN support will be improved and aligned to the requirements as per newly adopted RFC 8599. Besides the notification itself, we need to address some particularities in contact registration, derived from privacy concerns or device identification concerns.

noSQL adds-on

The DB layer needs a constant attention, so here is the plan for 3.1:

• add support for Dynamo noSQL DB, from Amazon, to improve the experience when comes to running OpenSIPS in AWS cloud.
• support for raw queries for Cassandra

https://blogopensips.files.wordpress.com/2019/12/opensips-3.1-crafting.jpg Routing calls and handling large volume of traffic is not a challenge anymore for OpenSIPS. The upcoming 3.1 release will focus on complex call crafting, basically on increasing OpenSIPS's ability to create and handle complex calling scenarios where multiple SIP calls are mixed and interacting. Or shortly said, the 3.1 release will address the Class 5 specific calling features and how to control such calling features via APIs.

As usual, all the OpenSIPS major releases are in depth presented during the OpenSIPS Summit yearly events. So, the 3.1 release is the star of OpenSIPS Summit in Amsterdam, May 2020.

Class 5 calling ingredients

Without actually using a Back2BAck model, but simply operating with calls (dialogs, UACs or UASs), many complex class 5 calling (and call mixing) scenarios may be scripted.

Calling API

A new OpenSIPS module, placed on top of dialog module, will allow remote control over the calls going through OpenSIPS. The module will expose an simplify set of command (API like) for setting up new calls, for answering and terminating calls, for transferring or putting on-hold calls - all these without interacting with the end-devices, but triggering and handling the action only from the OpenSIPS (as proxy) level.

Call mixing

Another new OpenSIPS module, that is able to manipulate calls/dialogs going through OpenSIPS, along with UAC/UAS dialogs (call originated or terminated into OpenSIPS), with the sole propose the mixing the RTP media between these multiple flows. The idea is to make possible the injection or stream of media from / to proxied calls, with the help of auxiliary calls initiated by OpenSIPS. Typical example is to play media within an existing call/dialog with nothing more than simple re-INVITEs - OpenSIPS will initial a new sip call to a media server, in order to get the RTP stream for the playback and it will push it into the proxied call by triggering in-dialog re-INVITEs.

Per-call hooks

As we already have for transactions, the dialog module will allow the script writer to set, in a per-dialog fashion, script routes to be triggered by various dialog events. Similar to t_on_failure(route), you can do dlg_on_timeout(route) to have a route called when the dialog lifetime is exceed. In the route, you may decide to prolong the lifetime, to terminate the call or do any other login. We can foreseen dlg_on_answer(route), dlg_on_terminate(route)' (and more) triggers, which will give a better interaction and control over the ongoing calls.

SDP topology hiding

Due to the specificity of class 5 scenarios, there is a real need to completely decouple the SDP's (not only from IP/port perspective) from the caller and callee side, like hiding the originator or overwriting the session name and version.

DTMF support

For both RTProxy and RTPEngine, OpenSIPS will be able to report to script the DTMF sampled from the passing RTP. This will make possible the implementation of simple IVRs or authentication via DTMF with nothing more than OpenSIPS and the media relay.

Extended BLF support

In Class 5 services, BLF is an important feature. Beside working out clustering support for BLF, an important task is reworking the BLF implementation to be call-branch aware, to be able to properly report the call events in parallel calling or call hunting scenarios.

Dialog module enhancements

We are looking at a good set of additions for the dialog module, like:

• improve the way of correlating multiple dialogs, and also to exchange data between calls (like accessing data specific to a call from a totally different call)
• sending in-dialog requests, crafted from script level
• better support for UAC or UAS like dialogs (not only proxy like)

Back-to-back support

The existing B2B implementation in OpenSIPS has some limitations, so we are looking to overcome via some major rework here.

Script driven B2B

Instead of using the XML scenario to drive the B2B logic (mixing between the calls), we want to use the OpenSIPS scripting for the purpose. This will eliminate all the limitation of the XML language (logic and action) and it will tremendously increase the level of integration of the B2B engine with the rest of the OpenSIPS functionalities. Shortly, more complex B2B logic will be possible, and also better integrated with the rest of OpenSIPS.

B2B clustering support

To be 100% production ready, an High-Availability support maybe available for the B2B engine. This will be achieved by adding clustering and replication support for the B2B calling.

B2B context

An important improvement of the B2B engine will be the addition of the B2B context, to help in correlating all the entities (UAC, UAS) part of a B2B session, to attach custom data to the entities and to exchange such data between the entities. This will help with Accounting and media relaying support for B2B, but also with building custom data sharing inside a B2B session (or between the sessions)

Call Center

For 3.1, we are looking at:

• adding clustering support and data replication for the call queue - this is extremely important for achieving High-Availability.
• more feature and metrics for managing the call queue
• distributed call-center or a goe-distributed single call queue which gets calls via different OpenSIPS instances and which distributed agents connected to different OpenSIPS instances.

Device Feature Key Synchronization (DFKS)

DFKS support is planned for OpenSIPS 3.1. This will help to keep feature settings in sync between multiple device and application servers, an essential need in a class 5 / PBX environment.

STIR/SHAKEN

https://blogopensips.files.wordpress.com/2018/12/opensips-3.1-icon.png For the upcoming OpenSIPS 3.1 release (and 3.x family) the main focus is on the devops concept. This translates into introducing and enhancing in OpenSIPS features / capabilities that (1) will increase the easiness when comes the writing / developing OpenSIPS scripts and (2) simplify the operational activities when comes to running and managing OpenSIPS.

This OpenSIPS 3.1 release is the star of OpenSIPS Summit in Amsterdam, April-May 2019 - beside presentations and workshops around the new cool things in this version, OpenSIPS 3.1 will also be the subject of several interactive demos on its new capabilities.

Script Development aspects

Generic Preprocessor Support

This feature adds full built-in pre-processing support for the OpenSIPS script. OpenSIPS 3.1 integrates various existing pre-processors within OpenSIPS. This simplify the scripting itself, the script portability across multiple servers and not to mention the entire deployment process of more complex platforms (where OpenSIPS is just a part of it). Even more, you will be able to use your preferred pre-processor and align OpenSIPS with the rest of your system (M4, Jinja, Embedded Ruby or others).
Read a full description of this feature here.

Module Functions Now Benefit From a New Parameter Interface

As a response to frequent mailing list complaints of wildly varying behaviors across different module functions (e.g. some accept integers/strings as inputs while others accept both integers/strings or variables holding such values), we've introduced an abstract layer which handles the parameter passing task for all module functions, effectively making all of them more powerful by globally allowing flexible input. An added benefit is that new OpenSIPS modules are now even faster to develop. See the new function calling conventions here.

Operational aspects

Routing Script Re-load

OpenSIPS 3.1 exposes the valuable ability of reloading the routes (not the module configuration) during runtime, with zero penalties and with zero loses as traffic. See the documentation of the MI "reload_routes" function.

Processes Auto-Scaling Support

This is the ability of OpenSIPS to scale up and down the number of processes at runtime. Basically OpenSIPS is able to automatically scale up (by forking new processes) according to the volume of traffic, or to scale down (terminating some worker processes) if the internal load is low. This means you do not have to worry if your estimation on the number for worker processes is correct or not (will my OpenSIPS hold to the traffic??) or to worry about planning restarts during the night (to manually resize the number of processes).
Read a full description of this feature here.

New OpenSIPS CLI (Command Line Interface) tool

Starting with OpenSIPS 3.1, the old opensipsctl tool becomes deprecated (as functionality and as software) and it is replaced by the new opensips-cli - a powerful Python3 application that allows you to interact in a smart way with OpenSIPS, to invoke advanced tools such as diagnose or tracer, as well as to perform DB provisioning. Read a full description of opensips-cli here.

Selectable Memory Allocator Support

This feature allows the internal memory manager to be selected at startup time. In OpenSIPS 3.1, the memory manager selection becomes a startup option, via command line arguments, allowing you to change it without any need to recompile or redeploy. Read a full description of this feature here.

Internal Memory Persistence during Restart

As there are several modules caching (in OpenSIPS internal memory, not in external no-sql cachers) large chunks of data, like Dynamic Routing, Dialplan, Dispatcher or Permissions, OpenSIPS 3.1 is able to avoid the date loading and caching penalty during a restart - this segments of the internal memory do "survive" during the restart. This dramatically reduces the time to restart of the entire service.
Read a full description of this feature here.

Unified Sharing Tags for Clustering

In 2.4, each module (with clustering support) is managing its own sharing tags completely isolated from other modules - this make the operating OpenSIPS a bit difficult sometime, as for a single switch from active to backup, you need to individually inform and change the tags in several modules, via several MI commands. In OpenSIPS 3.1 we have now the sharing tags managed by clusterer module itself and shared between multiple modules. So with a single MI command, changing a single sharing tag, you can control all the cluster-aware modules (like dialog timeouts, nathelper pinging, dispatcher pinging, etc) Read a full description of this feature here.

Integration aspects

MI Interaction Standardization

An extensive rework of the Management Interface in an attempt to standardize and speed up development of external applications which need to interact with OpenSIPS. The custom, JSON-based HTTP calls of OpenSIPS 2.X are now replaced with the JSON-RPC version 2 standard. The custom, line-oriented syntax was completely dropped. XMLRPC and mi_html (former mi_http) were kept backwards-compatible. Read a detailed description of the new interactions here

SMPP support

OpenSIPS 3.1 provides a new SMPP module that allows you to do bidirectional gatewaying between SIP and SMPP traffic - this is a powerful but flexible way to integrate with most of the SMS providers / gateways. Read a full description of this feature here.

RabbitMQ Consumer support

A new RabbitMQ consumer module which manages connections with one or more brokers, subscribes for events and exports them at OpenSIPS script level via the event interface.

But the full list of goodies offered by OpenSIPS 3.1 (and a more technical one too), together with migration instructions, can be found on the OpenSIPS 3.1 release notes page.

https://blogopensips.files.wordpress.com/2017/11/google-idi_018-1.jpg?w=300&h=200 The OpenSIPS 3.1 version is built around the clustering concept - today’s VoIP world is getting more and more dynamic, services are moving into Clouds and more and more flexibility is needed for the application to fully exploit such environments. But let’s pin point the main reasons for going for a clustered approach:

• scaling up with the processing/traffic load
• geographical distribution
• redundancy and High-Availability

the rest

The success of this release - in terms of achieving its integration goal - was strongly weighted by the collaboration with the teams of the partner projects, like SIPCapture, FreeSWITCH or CGRates. A collaboration in terms of ideas, brainstorming, solutions and of course, work. A collaboration that resulted in solutions and benefits for all the involved communities.

The best of

SIPCapture/Homer integration

The integration with the SIPCapture engine was a hot topic again. The work in this area focused in adding two new concepts (both on OpenSIPS and SIPCapture sides) when comes to capturing:

• non-SIP tracing - if up to this point, all the tracing was SIP-centric, now you can capture and visualize more types of data. You can capture information from transport layer (TCP, TLS, WS, WSS), information on the REST queries you performed from OpenSIPS script, information about the MI commands and also the script logs. All this information will help you (from operational perspective) to have a global view over how your OpenSIPS is doing (and lot limited to the SIP level only) - in other words, monitoring and troubleshooting will became much easier.
• data correlation - now that you have so many types of data traced, it is vital to be able to correlate them - to know what were the TCP/TLS/WSS connections involved in a SIP call, to know which were the REST queries or logs triggered by a call handling. Such correlation concept gives a new dimension to the tracing concept - you can navigate and jump between different data types in order to understand the relation between them (like why a SIP call failed by looking at the data from the transport level)

FreeSWITCH integration

As a powerful Class5 Engine, FreeSWITCH is the perfect complementary tool for OpenSIPS

About -> Available Versions -> 3.1.x Releases -> Release 3.1.0 Overview

OpenSIPS 3.1 philosophy

http://www.gearitservices.com/Portals/0/SitePics/SystemIntegration.jpg The OpenSIPS 3.1 version is built around the integration concept - the OpenSIPS ability to integrate and work together in all possible means with other projects, protocols, systems or concepts.
Why is integration so important to end up being the main tag of a major release? Well, everybody in the VoIP world is operating VoIP platforms/systems – and these are more than SIP Engines (as OpenSIPS is). Indeed, the SIP Engine is the core and most important part of the platform, but to build something usable and useful, you need additional components into your platform like CDR/billing engines, monitoring and tracing tools, data backends, non-SIP trunking or more specialized SIP engines. Shortly you need your SIP Engine (OpenSIPS, of course) to be able to easily integrate with all these components.

The success of this release - in terms of achieving its integration goal - was strongly weighted by the collaboration with the teams of the partner projects, like SIPCapture, FreeSWITCH or CGRates. A collaboration in terms of ideas, brainstorming, solutions and of course, work. A collaboration that

Integration

the rest

Migration from 2.2.x to 3.1.0?