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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.
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.
The following modules must be loaded before this module:
B2B_ENTITIES - Back-2-Back module used for handing server and client side calls.
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")
...
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));
}
...
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) | 91 | 58 | 2947 | 500 |
| 2. | Maksym Sobolyev (@sobomax) | 3 | 1 | 3 | 3 |
(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
Table 2.2. Most recently active contributors(1) to this module
| Name | Commit Activity | |
|---|---|---|
| 1. | Razvan Crainea (@razvancrainea) | May 2021 - Jul 2024 |
| 2. | Maksym Sobolyev (@sobomax) | Feb 2023 - Feb 2023 |
(1) including any documentation-related commits, excluding merge commits
Last edited by: Razvan Crainea (@razvancrainea).
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