B2B SDP De-Multiplexer Module
     __________________________________________________________

   Table of Contents

   1. Admin Guide

        1.1. Overview
        1.2. Use Cases
        1.3. Dependencies

              1.3.1. OpenSIPS Modules
              1.3.2. External Libraries or Applications

        1.4. Exported Parameters

              1.4.1. client_bye_mode (string)

        1.5. Exported Functions

              1.5.1. b2b_sdp_demux(URI[, [headers][, streams]])

   2. Contributors

        2.1. By Commit Statistics
        2.2. By Commit Activity

   3. Documentation

        3.1. Contributors

   List of Tables

   2.1. Top contributors by DevScore^(1), authored commits^(2) and
          lines added/removed^(3)

   2.2. Most recently active contributors^(1) to this module

   List of Examples

   1.1. Set client_bye_mode parameter
   1.2. Use b2b_sdp_demux() to handle an audio SIPREC call

Chapter 1. Admin Guide

1.1. Overview

   This module provides the logic to convert a multi-stream SDP
   call, to multiple calls, each containing a subset of streams
   from the initial call. The module only handles the SIP
   signalling part of the call, without interfering with the media
   of the call, which will flow end-to-end. The only manipulation
   it does is at the SDP level to disable the media-streams that
   are not being used downstream.

   The logic is implemented on top of the B2B module, and
   de-multiplexes a B2B server (the initial call with multiple
   streams) to multiple B2B clients (with their own streams
   subset). In-dialog requests that come from the initial caller
   will be forked towards each client, and their replies
   aggregated back to the caller. The other side in-dialog
   requests are forwarded to the caller as if only their stream
   had changed. When a call is terminated from the client side,
   the module can have different behaviors, according to the
   client_bye_mode parameter.

1.2. Use Cases

   A common scenario where this module can become useful is when
   configuring OpenSIPS as a SIPREC SRS proxy. Using this module
   you can receive on one side SIPREC INVITEs, which usually have
   two or more SDP streams (one for each call/conference
   participants), and split/de-multiplex each stream in a new call
   downstream, usually towards a media server that is able to do
   call recording. This way the media server will have to handle
   calls that contain a single media stream.

   Another use case is balancing multiple streams to different
   media servers. For example, if you are offering both audio and
   video services, you can split a two-stream call (with an audio
   and video stream) to two different calls, and send them to be
   processed by different servers. This way you may have separated
   audio-dedicated processing media servers, as well as
   video-dedicated one. Of course, this can be achieved if you can
   process the streams separately, for example for recording.

1.3. Dependencies

1.3.1. OpenSIPS Modules

   The following modules must be loaded before this module:
     * B2B_ENTITIES - Back-2-Back module used for handing server
       and client side calls.

1.3.2. External Libraries or Applications

   The following libraries or applications must be installed
   before running OpenSIPS with this module loaded:
     * None.

1.4. Exported Parameters

1.4.1. client_bye_mode (string)

   This parameter indicates how a BYE coming from the client side
   should be treated in the context of the upstream call.

   Possible values are:
     * disable - when a client terminates its call, the module
       will simply disable the media streams associated with its
       call, resulting in a re-INVITE upstream.
     * terminate - when one client terminates its call, the module
       will terminate all other calls, including the upstream one.
     * disable-terminate - same as disable, except that when the
       final stream is disabled, instead of a re-INVITE with all
       streams disabled, the module sends a BYE upstream.

   Default value is “disable”.

   Example 1.1. Set client_bye_mode parameter
...
modparam("b2b_sdp_demux", "client_bye_mode", "terminate")
...

1.5. Exported Functions

1.5.1.  b2b_sdp_demux(URI[, [headers][, streams]])

   Engages the B2B SDP De-Multiplexing scenario for the calls it
   has been triggered on.

   Parameters:
     * URI (string) - the URI where to send the newly generated
       calls
     * headers (AVP, optional) - an AVP containing multiple
       values, each index corresponding to one of the new calls
       generated by the function. The number of values in the AVP
       should be equal to the number of calls resulted, otherwise
       it may lead to an unexpected behavior. If missing, no extra
       headers will be added.
     * streams (AVP, optional) - an AVP containing multiple
       values, each value indicating the media stream index that
       should be used for the current client. If multiple streams
       should be used for a single call, they should be specified
       comma-separated (i.e. 0,2). The number of AVP values
       represent the number of calls generated downstream. If the
       parameter is missing, a call will be generated for each
       stream present in the initial call.

   This function can be used only from request route.

   Example 1.2. Use b2b_sdp_demux() to handle an audio SIPREC call
...
if (!has_totag() && is_method("INVITE")) {
        $avp(headers) = "X-Leg: caller\r\n");
        $avp(headers) = "X-Leg: callee\r\n");
        b2b_sdp_demux("sip:media@localhost", $avp(headers));
}
...

Chapter 2. Contributors

2.1. By Commit Statistics

   Table 2.1. Top contributors by DevScore^(1), authored
   commits^(2) and lines added/removed^(3)
                   Name               DevScore Commits Lines ++ Lines --
   1. Razvan Crainea (@razvancrainea)    93      59      2949     500
   2. Maksym Sobolyev (@sobomax)         4        2       5        5
   3. Alexandra Titoc                    4        2       3        2
   4. Norman Brandinger (@NormB)         3        1       1        1

   (1) DevScore = author_commits + author_lines_added /
   (project_lines_added / project_commits) + author_lines_deleted
   / (project_lines_deleted / project_commits)

   (2) including any documentation-related commits, excluding
   merge commits. Regarding imported patches/code, we do our best
   to count the work on behalf of the proper owner, as per the
   "fix_authors" and "mod_renames" arrays in
   opensips/doc/build-contrib.sh. If you identify any
   patches/commits which do not get properly attributed to you,
   please submit a pull request which extends "fix_authors" and/or
   "mod_renames".

   (3) ignoring whitespace edits, renamed files and auto-generated
   files

2.2. By Commit Activity

   Table 2.2. Most recently active contributors^(1) to this module
                   Name                 Commit Activity
   1. Razvan Crainea (@razvancrainea) May 2021 - Aug 2025
   2. Alexandra Titoc                 Sep 2024 - Sep 2024
   3. Norman Brandinger (@NormB)      Jun 2024 - Jun 2024
   4. Maksym Sobolyev (@sobomax)      Feb 2023 - Nov 2023

   (1) including any documentation-related commits, excluding
   merge commits

Chapter 3. Documentation

3.1. Contributors

   Last edited by: Razvan Crainea (@razvancrainea).

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