Pre-Submission Checklist

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Commits are split per component (core, individual modules, libs, utils, ...)
Each component has a single commit (if not, squash them into one commit)
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Type Of Change

New module

Checklist:

Tested with various sofphones : Ekiga, Linphone, CSipSimple

Description

By combining Kamailio, oRTP and mediastreamer2
this module is providing some very good foundation to support RTP endpoint and various media processing features.

This seems like a great idea for various use cases like IoT, IVR or other specific needs.

Kamailio is handling everything related to SIP/SDP (the module is adding some SDP parsing) as well as providing a scripting engine.

oRTP is providing RTP endpoints compliant with (RFC 3550)

MediaStreamer2, even if written in C is quite a high level library, because it is implementing a framework for audio processing using graphs of filters, filters can be to do various things.
Similar to playing with lego blocks :)

Support for most free and some non free codecs can be added easily.
Work to bridge calls is already in progress in the module.
Mobile phone support ARM CPU
other embedded scenario should be supported
much more ...

Mediastream2 is creating one thread per call "msticker", this can work smoothly in Kamailio even if it is forking processes.
Shared memory allocation is supported using wrapper around malloc/free used by the libraries.

Some extra work that needs to be done shorty

Syncronization with locks is not completed (minor task, only access to a linked list needs to be syncronized properly)
Memory leak not tested properly (not take the time to test with valgrind yet)

This project was started last year, I think it is time to submit it, I will surely find the time to do the extra work needed shortly.

Example config using the features already implemented

event_route[rms:start] {
        xnotice("[rms:start] play ...\n");
        rms_play("/tmp/reference_8000.wav", "rms:after_play");
};

event_route[rms:after_play] {
        xnotice("[rms:after_play] play done...\n");
        rms_hangup();
};

route {
        if (t_precheck_trans()) {
                t_check_trans();
                exit;
        }
        t_check_trans();
        if (is_method("INVITE") &amp;&amp; !has_totag()) {
                if (!rms_answer()) {
                        t_reply("503", "server error");
                }
        }

        if (is_method("BYE")){
                rms_media_stop();
        }
}

You can view, comment on, or merge this pull request online at:

https://github.com/kamailio/kamailio/pull/1701

Commit Summary

rtp_media_server: adding module

File Changes

M src/Makefile.groups (6)
A src/modules/rtp_media_server/Makefile (13)
A src/modules/rtp_media_server/config_example/kamailio.cfg (56)
A src/modules/rtp_media_server/doc/Makefile (4)
A src/modules/rtp_media_server/doc/rtp_media_server.xml (42)
A src/modules/rtp_media_server/doc/rtp_media_server_admin.xml (193)
A src/modules/rtp_media_server/install_bc.sh (45)
A src/modules/rtp_media_server/rms_media.c (320)
A src/modules/rtp_media_server/rms_media.h (103)
A src/modules/rtp_media_server/rms_sdp.c (301)
A src/modules/rtp_media_server/rms_sdp.h (45)
A src/modules/rtp_media_server/rtp_media_server.c (718)
A src/modules/rtp_media_server/rtp_media_server.h (91)
A src/modules/rtp_media_server/voice_file/Bach_10s_8000.wav (0)

Patch Links:

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