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About: Version-3-1-0

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

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Table of Content (hide)

  1. 1. Migration from 3.0.x to 3.1.0
  2. 2. What is new in 3.1.0
    1. 2.1 OpenSIPS script
    2. 2.2 OpenSIPS core
    3. 2.3 OpenSIPS core MI functions
    4. 2.4 [NEW] AUTH_JWT module
    5. 2.5 [NEW] Call Operations module
    6. 2.6 [NEW] Media Exchange module
    7. 2.7 [NEW] PRESENCE_DFKS module
    8. 2.8 [NEW] RATE_CACHER module
    9. 2.9 [NEW] STIR_SHAKEN module
    10. 2.10 [NEW] Quality-based Routing Module (qrouting)
    11. 2.11 [NEW] UUID module
    12. 2.12 ACC module
    13. 2.13 AUTH module
    14. 2.14 B2B_ENTITIES
    15. 2.15 B2B_LOGIC
    16. 2.16 CACHEDB_MONGODB module
    17. 2.17 CALL_CENTER module
    18. 2.18 CFGUTILS module
    19. 2.19 DIALOG module
    20. 2.20 DISPATCHER module
    21. 2.21 DROUTING module
    22. 2.22 EVENT_DATAGRAM module
    23. 2.23 Event RabbitMQ module
    24. 2.24 FRAUD_DETECTION module
    25. 2.25 LOAD_BALANCER
    26. 2.26 MATHOPS module
    27. 2.27 MI_FIFO module
    28. 2.28 MID_REGISTRAR module
    29. 2.29 PUA_DIALOGINFO module
    30. 2.30 PRESENCE module
    31. 2.31 PROTO_SMPP module
    32. 2.32 REGISTRAR module
    33. 2.33 REST_CLIENT module
    34. 2.34 RTPEngine module
    35. 2.35 RTPProxy module
    36. 2.36 SIGNALING module
    37. 2.37 SIPREC module
    38. 2.38 SMPP module
    39. 2.39 TOPOLOGY_HIDING module
    40. 2.40 TLS_MGM module
    41. 2.41 UAC_AUTH module
    42. 2.42 USRLOC module

1.  Migration from 3.0.x to 3.1.0


2.  What is new in 3.1.0

2.1  OpenSIPS script

2.2  OpenSIPS core

2.3  OpenSIPS core MI functions

2.4  [NEW] AUTH_JWT module

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.

2.5  [NEW] Call Operations module

The new CallOps module provides a set of functions to control the behavior of an ongoing call. Using this module you can start a new call, trigger call blind and attended transfers, put a call on hold, etc. Check out more information in the module's documentation page.

2.6  [NEW] Media Exchange module

A new module that provides means to exchange SDP bodies between different calls has been added. Using this new module one can implement enhanced features such as playing back announcements or music on hold files during an ongoing call, or listening a conversation of two different participants. You can find more information about the new module in the module's documentation page.

2.7  [NEW] PRESENCE_DFKS module

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. Technical specs and more details are available in the module's readme. Also, for more , see this blog post.

2.8  [NEW] RATE_CACHER module

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.

2.9  [NEW] STIR_SHAKEN module

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.

2.10  [NEW] Quality-based Routing Module (qrouting)

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)

2.11  [NEW] UUID module

This module provides a way to generate universally unique identifiers (UUID) as specified in RFC 4122 for use in the OpenSIPS script.

2.12  ACC module

2.13  AUTH module

2.14  B2B_ENTITIES

2.15  B2B_LOGIC

2.16  CACHEDB_MONGODB module

2.17  CALL_CENTER module

2.18  CFGUTILS module

2.19  DIALOG module

2.20  DISPATCHER module

2.21  DROUTING module

2.22  EVENT_DATAGRAM module

2.23  Event RabbitMQ module

2.24  FRAUD_DETECTION module

2.25  LOAD_BALANCER

2.26  MATHOPS module

2.27  MI_FIFO module

2.28  MID_REGISTRAR module

2.29  PUA_DIALOGINFO module

There is major rework of the module not make it call forking enabled - the 'pua_dialoginfo' module is able now to get access to the transaction information (not only to the dialog information as before); this made the module aware of the multiple branches created, not only dialog aware, so data related to call branches can now be also published. The new BLF implementation has two major advantages:

See more details in this comprehensive blog post.

2.30  PRESENCE module

2.31  PROTO_SMPP module

2.32  REGISTRAR module

2.33  REST_CLIENT module

2.34  RTPEngine module

2.35  RTPProxy module

2.36 SIGNALING module

2.37  SIPREC module

2.38  SMPP module

2.39  TOPOLOGY_HIDING module

2.40  TLS_MGM module

2.41  UAC_AUTH module

2.42  USRLOC module

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Page last modified on June 02, 2020, at 03:39 PM