# &adminguide;

## Overview

rtp_media_server module adds RTP and media processing functionalities to Kamailio. Kamailio is providing SIP signaling including an endpoint with Dialog state, SDP parsing and scripting language.

oRTP: is providing Real-time Transport Protocol (RFC 3550)

mediastreamer2: is providing mediaprocessing functionnalities using graphs and filters, many modules are available to support various features, it should be relatively simple to integrated them.

mediastreamer2 is also providing a framework to create custom mediaprocessing modules.

## Quick start, how-to build on debian

### Building on Debian, Dockerfile and docker image

The module includes Dockerfile that can also be used as a reference on how to build everything from source on Debian, the package of libmediastreamer on Linux is usually outdated.

A Docker image is also available from dockerhub https://hub.docker.com/r/jchavanton/rtp_media_server

## Dependencies

### &kamailio; Modules

The module depends on the following modules (in other words the listed modules must be loaded before this module):

- tm - transaction module

### External Libraries or Applications

The following libraries or applications must be installed before running &kamailio; with this module loaded:

If you want to build oRTP and mediastreamer from source, you can use the provided script for Debian "install_bc.sh".

- oRTP git clone git://git.linphone.org/ortp.git

  oRTP is a library implementing Real-time Transport Protocol (RFC 3550), distributed under GNU GPLv2 or proprietary license.

- mediastreamer2 git clone git://git.linphone.org/mediastreamer2.git

  Mediastreamer2 is a powerful and lightweight streaming engine specialized for voice/video telephony applications.

- bcunit git clone https://github.com/BelledonneCommunications/bcunit.git

  fork of the defunct project CUnit, with several fixes and patches applied. CUnit is a Unit testing framework for C.

## Parameters

### `log_file_name` (string)

oRTP and MediaStreamer2 log file settings the log mask is not configurable : MESSAGE | WARNING | ERROR | FATAL levels are activated.

Default value is not-set (no logging to file).

#### log_file_name example

    ...
    modparam("rtp_media_server", "log_file_name", "/var/log/rms/rms_ortp.log")
    ...

## Functions

### `rms_answer` (event_route)

Create a call leg with a SIP dialog and an RTP session call the event_route

This function can be used from REQUEST_ROUTE, REPLY_ROUTE and FAILURE_ROUTE.

#### rms_answer usage example

    ...
    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") && !has_totag()) {
    		if (!rms_answer("rms:start")) {
    			t_reply("503", "server error");
    		}
    	}
    	rms_sip_request();
    ...

### `rms_hangup` ()

Send a BYE, delete the RTP session and the media resources.

This function can be used from EVENT_ROUTE.

#### rms_hangup usage example

    ...
    	rms_hangup();
    ...

### `rms_bridge` (target URI, event_route)

Bridge the incoming call, create a second call leg using a UAC in a B2BUA manner, this is needed in case we want to un-bridge later, a feature not currently implemented Call the specified event_route, defaulting to [rms:bridge].

This function can be used from REQUEST_ROUTE.

#### rms_bridge usage example

    ...
    event_route[rms:bridged] {
    	xnotice("[rms:bridged] !\n");
    };

    route {
    	if (t_precheck_trans()) {
    		t_check_trans();
    		exit;
    	}
    	t_check_trans();
    	if (is_method("INVITE") && !has_totag()) {
    		$var(target) = "sip:" + $rU + "@mydomain.com:5060;";
    		if (!rms_bridge("$var(target)", "rms:bridged")) {
    			t_reply("503", "server error");
    		}
    	}
    	if(rms_dialog_check()) // If the dialog is managed by the RMS module, the in-dialog request needs to be handled by it.
    		rms_sip_request();
    ...

### `rms_dialog_check` ()

Returns true if the current SIP message it handled/known by the RMS module, else it may be handled in any other way by Kamailio.

This function can be used from REQUEST_ROUTE, REPLY_ROUTE and FAILURE_ROUTE.

#### rms_dialog_check usage example

    ...
    	if (rms_dialog_check()) {
    		xnotice("This dialog is handled by the RMS module\n");
    		rms_sip_request();
    	}
    ...

### `rms_sip_request` ()

This should be called for every in-dialog SIP request, it will be forwarded behaving as a B2BUA, the transaction will be suspended until the second leg replies.

If the SIP dialog is not found "481 Call/Transaction Does Not Exist" is returned.

This function can be used from REQUEST_ROUTE, REPLY_ROUTE and FAILURE_ROUTE.

#### rms_sip_request usage example

    ...
    	if (rms_dialog_check()) {
    		rms_sip_request();
    	}
    ...

### `rms_play` (file, event_route)

Play a wav file, a resampler is automatically configured to resample and convert stereo to mono if needed.

The second parameter is the event route that will be called when the file was played.

This function can be used from EVENT_ROUTE.

#### rms_play usage example

    ...
    	rms_play("file.wav", "event_route_name");
    ...