Right now the wolfSSL code in directly imported into the OpenSIPS code. Why ? more or less for the same reason as we needed the openSSL twitches. The wolfSSL needs special compiling flags in order to work in a multi-process environment. This means we are stuck to a certain verions, without easy way to update to latest releases of wolfSSL. Thanks to the TCP rework and moving to a multi-threaded architecture (for TCP engine), we will be able to dynamically link to the wolfssl library, without any special twitches. So, we will have access to the latest versions, on different distros.

To see how this works, check out this [[https://blog.opensips.org/2026/03/25/sip-prack-interworking-support-in-opensips-4-0/|detailed blog post]].

.

!!!! Proxy Protocol

Starting with OpenSIPS 4.0, we introduced support for the Proxy Protocol, allowing OpenSIPS to transparently receive and forward the original client addressing information even when connections pass through intermediate proxies.
The Proxy Protocol was originally developed by HAProxy Technologies for HAProxy, as a lightweight method for proxies and load balancers to forward the original client IP address and port to backend servers.
To see how this works, check out this [[https://blog.opensips.org/2026/02/24/proxy-protocol-support-in-opensips-4-0/|detailed blog post]].

So we need to be able to have more, but without wasting memory - and this is what 4.0 does - and to see how is done, check out this [[https://blog.opensips.org/2026/02/17/big-scale-sip-forking/|detailed blog post].

The solution here is to create '''bond''' interfaces, groups of sockets to serve a logical destination (like "towards carriers" or "towards internal" or "towards customers") and contain a listing of real sockets with various protocols and AF's. OpenSIPS will automatically pick from the group the socket matching the final proto:ip:AF:port coordinates. A bond socket/interface can be used at script level as any other socket.

%rframe width=300px % https://blogopensips.files.wordpress.com/2023/11/opensips-4.0-ims.jpg

Bits and pieces of the '''IMS (IP Multimedia Subsystem)''' topic were part of the previous OpenSIPS release, but this 4.0 fully focused on the IMS part. Considering the traction and need of IMS solutions, the implementation of a consistent and large IMS support in OpenSIPS was a mandatory step in order to answer to the needs of the industry.

The IMS topic is a large one, transcending the SIP protocol and the scope of OpenSIPS. So, the OpenSIPS 4.0 worked out the parts of the IMS ecosystem which are based or related to SIP, of course.
\\
\\

!!! IMS

%rframe width=300px % https://blogopensips.files.wordpress.com/2023/11/opensips-4.0-ims-dia.png

The IMS topic is a very large and complex one. And the goal here was to design and implement an IMS support that is truly able to provide solutions for the industry. So, a quite extensive stage of exploring, understanding and designing on the IMS ecosystem was required in a first stage.\\

This work was undertaken by the [[https://www.opensips.org/Development/Working-Groups#ims-owg|'''IMS OpenSIPS Working Group (IMS OWG)''']], a gathering of people with common inters in IMS, working together to draft, design and implement the IMS support in OpenSIPS.\\

The IMS Working group compiled a [[https://github.com/OpenSIPS/opensips/wiki/IMS-Working-Group-Scope|Scope Of Work]], which helped in laying down the [[https://github.com/OpenSIPS/opensips/wiki/IMS-Working-Group-Requirements|actual requirements]]. The group decision was to tackle, at this first stage, '''the CSCF components, together with its interfaces'''.\\

As results, this translated in work across several module along with the addition of new specialized modules.

!!!! AKA digest authentication

Present in the '''S-CSCF''', the AKA authentication mechanism (RFC 3310) is implemented by the new [[https://opensips.org/docs/modules/4.0.x/auth_aka.html|AUTH_AKA]] module. Its complex interaction through the Cx/Dx Diameter interface is provided by the new [[https://opensips.org/docs/modules/4.0.x/aka_av_diameter.html|AKA_AV_DIAMETER]] module, responsible for fetching and managing the authentication vectors required by AKA.

!!!! IPSEC support

Present in the '''P-CSCF'''

!!!! AAA support

!!!! Presence support


!!! SQL operations

!!! Launch Darkly

!!! Message Queue

!!! Miscellaneous but important

The IMS Working group compiled a [[https://github.com/OpenSIPS/opensips/wiki/IMS-Working-Group-Scope|Scope Of Work]], which helped in laying down the [[https://github.com/OpenSIPS/opensips/wiki/IMS-Working-Group-Requirements|actual requirements]]. The group decision was to tackle, at this first stage, the CSCF components, together with its interfaces.


!!!!

!!!!

This task was undertaken by the [[https://www.opensips.org/Development/Working-Groups#ims-owg|'''IMS OpenSIPS Working Group (IMS OWG)''']], a gathering of people with common inters in IMS, working together to draft, design and implement the IMS support in OpenSIPS.

Various topics were addressed by the past releases, but most of the work with regards to each topic was mainly covered within a single release, so limited in terms of allocated resources (time and man-power). And some of these past topics are actually wider than what we managed to deliver. This is why the OpenSIPS 4.0 addressed the '''consolidation''' of various past topics, to complete or even expand them.
But one of the most important sides of this consolidation process is the '''testing of OpenSIPS, from the performance and conformity perspective'''. This testing was a long overdue task that definitely needs to be addressed.

!!! B2B enhancements

!!! SIPssert testing engine

!!! Performance and profiling testing

!!! Logging

!!!
!!! More Media relaying

%rframe width=250px % https://blogopensips.files.wordpress.com/2021/12/opensips-4.0-im.jpg
The instant messaging is evolving and gaining more and more importance. This happens in both worlds of VoIP/ITSP and IMS/MNO/MVNOs - and the overlapping and mixing of these worlds in increasing each year. The OpenSIPS 4.0 is to address more, in depth, the topic of '''Instant Messaging''', in the context of SIP. And this is done through two major perspectives: from the '''Unified Communication''' perspective, by aligning IM with the voice and presence capabilities in OpenSIPS, and from the '''IMS''' perspective, mainly as support/integration with RCS (Rich Communication Services).

Learn more from this [[https://blog.opensips.org/2022/01/20/opensips-4-0-messaging-in-the-ims-and-uc-ecosystems/|blog post]] about the OpenSIPS 4.0's vision on Instant Messaging in the IMS and Unified Communication ecosystem
\\
\\

!!! Messaging sessions or MSRP

!!!! Clustering engine

!!! Unified communication

!!! IMS

!!! Status Report support

!!! TCP boost

%rframe width=300px % https://blogopensips.files.wordpress.com/2019/12/opensips-4.0-crafting.jpg
Routing calls and handling large volume of traffic is not a challenge anymore for OpenSIPS. The upcoming 4.0 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 4.0 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 4.0 release is the star of [[http://www.opensips.org/events/Summit-2020Amsterdam/|OpenSIPS Summit in Amsterdam, May 2020]].

----

[[#class5]]
!!! 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 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.

!!!! 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 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.

!!!! 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 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.

!!!! 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 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.

!!!! Extended BLF support
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'''.

!!!! 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)

----

[[#b2b]]
!!! 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 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.

!!!! 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).

----

[[#callcenter]]
!!! Call Center
For 4.0, 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 geo-distributed single call queue which gets calls via different OpenSIPS instances and which distributed agents connected to different OpenSIPS instances.

----

[[#DFKS]]
!!! Device Feature Key Synchronization (DFKS)
[[http://community.polycom.com/polycom/attachments/polycom/VoIP/2234/1/DeviceFeatureKeySynchronizationFD.pdf|DFKS support]] is planned for OpenSIPS 4.0. 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]]
!!! STIR/SHAKEN
Dealing with Robocalling and CLI spoofing becomes a must when building advanced calling solutions. The [[https://www.slideshare.net/slideshow/embed_code/key/Ly9vOqcKuML5Nq|support for STIR/SHAKEN]] will be part of OpenSIPS 4.0, 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).

----

[[#PN]]
!!! Push Notification (RFC8599)
The existing PN support will be improved and aligned to the requirements as per newly adopted [[https://tools.ietf.org/html/rfc8599|RFC 8599]]. Besides the notification itself, we need to address some particularities in contact registration, derived from privacy concerns or device identification concerns.

----

[[#DB]]
!!! noSQL adds-on
The DB layer needs a constant attention, so here is the plan for 4.0:
* add support for [[https://aws.amazon.com/dynamodb/| Dynamo]] noSQL DB, from Amazon, to improve the experience when comes to running OpenSIPS in AWS cloud.
* support for raw queries for Cassandra

----

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.

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.

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.

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'''.

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.

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)

!!!! B2B engine
Improve the media

* 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.

Dealing with Robocalling and CLI spoofing becomes a must when building advanced calling solutions. The [[https://www.slideshare.net/slideshow/embed_code/key/Ly9vOqcKuML5Nq|support for STIR/SHAKEN]] will be part of OpenSIPS 4.0, 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).

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 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.

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.

Routing calls and handling large volume of traffic is not a challenge anymore for OpenSIPS. The upcoming 4.0 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 4.0 release will address the Class 5 specific calling features and how to control such calling features via APIs.

%rframe width=300px % https://blogopensips.files.wordpress.com/2018/12/opensips-4.0-icon.png
For the upcoming OpenSIPS 4.0 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 4.0 release is the star of [[http://www.opensips.org/events/Summit-2019Amsterdam/ | OpenSIPS Summit in Amsterdam, April-May 2019 ]] - beside presentations and workshops around the new cool things in this version, OpenSIPS 4.0 will also be the subject of several interactive demos on its new capabilities.


----

[[#development]]
!!! Script Development aspects

!!!! Generic Preprocessor Support

This feature adds full built-in pre-processing support for the OpenSIPS script. OpenSIPS 4.0 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 [[https://blog.opensips.org/2019/03/05/generic-preprocessor-support-in-opensips-4-0/|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 [[https://www.opensips.org/Documentation/Script-Syntax-4-0#function-calling-conventions|here]].

----
[[#operational]]
!!! Operational aspects

!!!! Routing Script Re-load

OpenSIPS 4.0 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 [[https://www.opensips.org/Documentation/Interface-CoreMI-4-0#toc8|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 [[https://blog.opensips.org/2019/02/25/auto-process-scaling-a-cure-for-load-and-resources-concerns/|here]].

!!!! New OpenSIPS CLI (Command Line Interface) tool

Starting with OpenSIPS 4.0, 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 [[https://github.com/OpenSIPS/opensips-cli/blob/master/docs/modules/diagnose.md|diagnose]] or [[https://github.com/OpenSIPS/opensips-cli/blob/master/docs/modules/tracer.md|tracer]], as well as to perform DB provisioning.
Read a full description of ''opensips-cli'' [[https://blog.opensips.org/2019/03/13/new-opensips-cli-tool-for-the-new-management-interface-in-opensips-4-0/|here]].

!!!! Selectable Memory Allocator Support

This feature allows the internal memory manager to be selected at startup time. In OpenSIPS 4.0, 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 [[https://blog.opensips.org/2019/03/21/containing-memory-related-issues-in-an-easy-way-with-opensips-4-0/|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 4.0 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 [[https://blog.opensips.org/2019/04/04/no-downtime-for-opensips-4-0-restarts/|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 4.0 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 [[https://blog.opensips.org/2019/03/07/achieving-service-redundancy-in-two-steps-with-unified-clustering-in-opensips-4-0/|here]].


----

[[#integration]]
!!! 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 [[https://www.opensips.org/Documentation/Interface-MI-4-0#protocols|here]]

!!!! SMPP support

OpenSIPS 4.0 provides a [[https://opensips.org/html/docs/modules/4.0.x/proto_smpp.html|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 [[https://blog.opensips.org/2019/01/24/gateway-between-sip-and-smpp-messages/|here]].

!!!! RabbitMQ Consumer support

A new [[https://opensips.org/html/docs/modules/4.0.x/rabbitmq_consumer.html|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 4.0 (and a more technical one too), together with migration instructions, can be found on the [[https://www.opensips.org/About/Version-4-0-0|OpenSIPS 4.0 release notes page]].

%rframe% https://blogopensips.files.wordpress.com/2017/11/google-idi_018-1.jpg?w=300&h=200
The OpenSIPS 4.0 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 best of

!!!! SIPCapture/Homer integration


!!!! FreeSWITCH integration

As a powerful Class5 Engine, [[http://www.freeswitch.org|FreeSWITCH]] is the perfect complementary tool for OpenSIPS

!!!! the rest




!!! [[Documentation/Migration-2-2-0-to-4-0-0|Migration from 2.2.x to 4.0.0]]

%rframe% http://www.gearitservices.com/Portals/0/SitePics/SystemIntegration.jpg
The OpenSIPS 4.0 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 [[http://www.sipcapture.org|SIPCapture]], [[http://www.freeswitch.org|FreeSWITCH]] or [[http://cgrates.org/|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 integration with the [[http://www.sipcapture.org|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)

!!! 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




!!! Integration