# &adminguide;

## Overview

This module is used by &osips; to communicate with RTPProxy, a media relay proxy used to make the communication between user agents behind NAT possible.

This module is also used along with RTPProxy to record media streams between user agents or to play media to either &ua;c or &ua;s.

## Multiple RTPProxy usage

Currently, the rtpproxy module can support multiple rtpproxies for balancing/distribution and control/selection purposes.

The module allows the definition of several sets of rtpproxies - load-balancing will be performed over a set and the user has the ability to choose what set should be used. The set is selected via its id - the id being defined along with the set. Refer to the rtpproxy_sock module parameter definition for syntax description.

The balancing inside a set is done automatically by the module based on the weight of each rtpproxy from the set. Note that if rtpproxy has weight 0, it will be used only when no other rtpproxies (with a different weight value than 0) respond. Default weight is 1.

Starting with &osips; 1.11, the set_rtp_proxy_set() function has been removed. The set is now specified for each function. If absend, the default set 0 is used. Also, engage_rtp_proxy(), unforce_rtp_proxy() and start_recording() functions have been deprecated and replaced by rtpproxy_engage(), rtpproxy_unforce() and rtpproxy_start_recording() respectively.

IMPORTANT: if you use multiple sets, make sure you use the same set for both rtpproxy_offer()/rtpproxy_answer() and rtpproxy_unforce()!!

## RTPProxy timeout notifications

Nathelper module can also receive timeout notifications from multiple rtpproxies. RTPProxy can be configured to send notifications when a session doesn't receive any media for a configurable interval of time. The rtpproxy modules has implemented a listener for such notifications and when received it terminates the dialog at SIP level (send BYE to both ends), with the help of dialog module.

In our tests with RTPProxy we observed some limitations and also provide a patch for it against git commit 600c80493793bafd2d69427bc22fcb43faad98c5. It contains an addition and implements separate timeout parameters for the phases of session establishment and ongoing sessions. In the official code a single timeout parameter controls both session establishment and rtp timeout and the timeout notification is also sent in the call establishment phase. This is a problem since we want to detect rtp timeout fast, but also allow a longer period for call establishment.

To enable timeout notification there are several steps that you must follow:

Start &osips; timeout detection by setting the rtpp_notify_socket module parameter in your configuration script. This is the socket where further notification will be received from rtpproxies. This socket must be a TCP or UNIX socket. Also, for all the calls that require notification, the rtpproxy_engage(), rtpproxy_offer() and rtpproxy_answer() functions must be called with the n flag.

Configure RTPProxy to use timeout notification by adding the following command line parameters:

- -n timeout_socket - specifies where the notifications will be sent. This socket must be the same as rtpp_notify_socket &osips; module parameter. This parameter is mandatory.

- -T ttl - limits the rtp session timeout to ttl. This parameter is optional and the default value is 60 seconds.

- -W ttl - limits the session establishment timeout to ttl. This parameter is optional and the default value is 60 seconds.

All of the previous parameters can be used with the offical RTPProxy release, except for the last one. It has been added, together with other modifications to RTPProxy in order to work properly. The patch is located in the patches directory in the module.

To get the patched version from git you must follow theese steps:

- Get the latest source code: git clone git://sippy.git.sourceforge.net/gitroot/sippy/rtpproxy

- Make a branch from the commit: git checkout -b branch_name 600c80493793bafd2d69427bc22fcb43faad98c5

- Patch RTPProxy: patch < path_to_rtpproxy_patch

The patched version can also be found at: http://opensips.org/pub/rtpproxy/

## Dependencies

### &osips; Modules

The following modules must be loaded before this module:

- a database module - only if you want to load use a database table from where to load the rtp proxies sets.

- dialog module - if using the rtpproxy_engage functions or RTPProxy timeout notifications.

### External Libraries or Applications

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

- None.

## Exported Parameters

### `rtpproxy_sock` (string)

Definition of socket(s) used to connect to (a set) RTPProxy. It may specify a UNIX socket or an IPv4/IPv6 UDP socket.

Default value is NONE (disabled).

#### Set `rtpproxy_sock` parameter

    ...
    # single rtproxy with specific weight
    modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:12221=2")
    # multiple rtproxies for LB
    modparam("rtpproxy", "rtpproxy_sock",
    	"udp:localhost:12221 udp:localhost:12222")
    # multiple sets of multiple rtproxies
    modparam("rtpproxy", "rtpproxy_sock",
    	"1 == udp:localhost:12221 udp:localhost:12222")
    modparam("rtpproxy", "rtpproxy_sock",
    	"2 == udp:localhost:12225")
    ...

### `rtpproxy_disable_tout` (integer)

Once RTPProxy was found unreachable and marked as disable, rtpproxy will not attempt to establish communication to RTPProxy for rtpproxy_disable_tout seconds.

Default value is 60.

#### Set `rtpproxy_disable_tout` parameter

    ...
    modparam("rtpproxy", "rtpproxy_disable_tout", 20)
    ...

### `rtpproxy_timeout` (string)

Timeout value in waiting for reply from RTPProxy.

Default value is 1.

#### Set `rtpproxy_timeout` parameter to 200ms

    ...
    modparam("rtpproxy", "rtpproxy_timeout", "0.2")
    ...

### `rtpproxy_autobridge` (integer)

Enable auto-bridging feature. Does not properly function when doing serial/parallel forking!

Default value is 0.

#### Enable auto-bridging feature

    ...
    modparam("rtpproxy", "rtpproxy_autobridge", 1)
    ...

### `rtpproxy_tout` (integer)

Obsolete. see rtpproxy_timeout.

### `rtpproxy_retr` (integer)

How many times rtpproxy should retry to send and receive after timeout was generated.

Default value is 5.

#### Set `rtpproxy_retr` parameter

    ...
    modparam("rtpproxy", "rtpproxy_retr", 2)
    ...

### `nortpproxy_str` (string)

The parameter sets the SDP attribute used by rtpproxy to mark the packet SDP informations have already been mangled.

If empty string, no marker will be added or checked.

The string must be a complete SDP line, including the EOH (\r\n).

Default value is a=nortpproxy:yes\r\n.

#### Set `nortpproxy_str` parameter

    ...
    modparam("rtpproxy", "nortpproxy_str", "a=sdpmangled:yes\r\n")
    ...

### `db_url` (string)

The database url. This parameter should be set if you want to use a database table from where to load or reload definitions of socket(s) used to connect to (a set) RTPProxy. The record from the database table will be read at start up (added to the ones defined with the rtpproxy_sock module parameter) and when the MI command rtpproxy_reload is issued(the definitions will be replaced with the ones from the database table).

Default value is NULL.

#### Set `db_url` parameter

    ...
    modparam("rtpproxy", "db_url",
    		"mysql://opensips:opensipsrw@192.168.2.132/opensips")
    ...

### `db_table` (string)

The name of the database table containing definitions of socket(s) used to connect to (a set) RTPProxy.

Default value is rtpproxy_sockets.

#### Set `db_table` parameter

    ...
    modparam("rtpproxy", "db_table", "nh_sockets")
    ...

### `rtpp_socket_col` (string)

The name rtpp socket column in the database table.

Default value is rtpproxy_sock.

#### Set `rtpp_socket_col` parameter

    ...
    modparam("rtpproxy", "rtpp_socket_col", "rtpp_socket")
    ...

### `set_id_col` (string)

The name set id column in the database table.

Default value is set_id.

#### Set `set_id` parameter

    ...
    modparam("rtpproxy", "set_id_col", "rtpp_set_id")
    ...

### `rtpp_notify_socket` (string)

The socket used by &osips; to receive timeout notifications.

Default value is NULL.

#### Set `rtpp_notify_socket` parameter

    ...
    modparam("rtpproxy", "rtpp_notify_socket", "tcp:10.10.10.10:9999")
    ...

## Exported Functions

### `engage_rtp_proxy([flags [, ip_address [, set_id [, sock_pvar]]]]) - deprecated`, `rtpproxy_engage([flags [, ip_address [, set_id [, sock_pvar]]]])`

Rewrites &sdp; body to ensure that media is passed through an &rtp; proxy. It uses the dialog module facilities to keep track when the rtpproxy session must be updated. Function must only be called for the initial INVITE and internally takes care of rewriting the body of 200 OKs and ACKs. Note that when used in bridge mode, this function might advertise wrong interfaces in &sdp; (due to the fact that &osips; is not aware of the RTPProxy configuration), so you might face an undefined behavior.

Meaning of the parameters is as follows:

- flags(optional) - flags to turn on some features.

  - a - flags that UA from which message is received doesn't support symmetric RTP.

  - l - force lookup, that is, only rewrite SDP when corresponding session is already exists in the RTP proxy. By default is on when the session is to be completed (reply in non-swap or ACK in swap mode).

  - i/e - when RTPProxy is used in bridge mode, these flags are used to indicate the direction of the media flow for the current request/reply. 'i' refers to the LAN (internal network) and corresponds to the first interface of RTPProxy (as specified by the -l parameter). 'e' refers to the WAN (external network) and corresponds to the second interface of RTPProxy. These flags should always be used together. For example, an INVITE (offer) that comes from the Internet (WAN) to goes to a local media server (LAN) should use the 'ei' flags. The answer should use the 'ie' flags. Depending on the scenario, the 'ii' and 'ee' combination are also supported. Only makes sense when RTPProxy is running in the bridge mode.

  - f - instructs rtpproxy to ignore marks inserted by another rtpproxy in transit to indicate that the session is already goes through another proxy. Allows creating chain of proxies.

  - r - flags that IP address in SDP should be trusted. Without this flag, rtpproxy ignores address in the SDP and uses source address of the SIP message as media address which is passed to the RTP proxy.

  - o - flags that IP from the origin description (o=) should be also changed.

  - c - flags to change the session-level SDP connection (c=) IP if media-description also includes connection information.

  - s/w - flags that for the UA from which message is received, support symmetric RTP must be forced.

  - n - flags that enables the notification timeout for the session.

  - zNN - requests the RTPproxy to perform re-packetization of RTP traffic coming from the UA which has sent the current message to increase or decrease payload size per each RTP packet forwarded if possible. The NN is the target payload size in ms, for the most codecs its value should be in 10ms increments, however for some codecs the increment could differ (e.g. 30ms for GSM or 20ms for G.723). The RTPproxy would select the closest value supported by the codec. This feature could be used for significantly reducing bandwith overhead for low bitrate codecs, for example with G.729 going from 10ms to 100ms saves two thirds of the network bandwith.

- ip_address(optional) - new SDP IP address.

- set_id(optional) - the set used for this call.

- sock_pvar(optional) - pvar used to store the RTPProxy socket chosen for this call. Note that the variable will only be populated in the initial request.

This function can be used from REQUEST_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE.

#### `rtpproxy_engage` usage

    ...
    if (is_method("INVITE") && has_totag()) {
    	if ($var(setid) != 0) {
    		rtpproxy_engage(,,"$var(setid)", "$var(proxy)");
    		xlog("SCRIPT: RTPProxy server used is $var(proxy)\n");
    	} else {
    		rtpproxy_engage();
    		xlog("SCRIPT: using default RTPProxy set\n");
    	}
    }
    ...

### `rtpproxy_offer([flags [, ip_address [, set_id [, sock_pvar]]]])`

Rewrites &sdp; body to ensure that media is passed through an &rtp; proxy. To be invoked on INVITE for the cases the SDPs are in INVITE and 200 OK and on 200 OK when SDPs are in 200 OK and ACK.

See rtpproxy_engage() function description above for the meaning of the parameters.

This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE.

#### `rtpproxy_offer` usage

    route {
    ...
        if (is_method("INVITE")) {
            if (has_body("application/sdp")) {
                if (rtpproxy_offer())
                    t_on_reply("1");
            } else {
                t_on_reply("2");
            }
        }
        if (is_method("ACK") && has_body("application/sdp"))
            rtpproxy_answer();
    ...
    }

    onreply_route[1]
    {
    ...
        if (has_body("application/sdp"))
            rtpproxy_answer();
    ...
    }

    onreply_route[2]
    {
    ...
        if (has_body("application/sdp"))
            rtpproxy_offer();
    ...
    }

### `rtpproxy_answer([flags [, ip_address [, set_id [, sock_pvar]]]])`

Rewrites &sdp; body to ensure that media is passed through an &rtp; proxy. To be invoked on 200 OK for the cases the SDPs are in INVITE and 200 OK and on ACK when SDPs are in 200 OK and ACK.

See rtpproxy_engage() function description above for the meaning of the parameters.

This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE.

#### `rtpproxy_answer` usage

See rtpproxy_offer() function example above for example.

### `unforce_rtp_proxy([set_id [, sock_pvar]]) - deprecated`, `rtpproxy_unforce([set_id [, sock_pvar]])`

Tears down the RTPProxy session for the current call.

Meaning of the parameters is as follows:

- set_id(optional) - the set used for this call.

- sock_pvar(optional) - pvar used to store the RTPProxy socket chosen for this call.

This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE.

#### `rtpproxy_unforce` usage

    ...
    rtpproxy_unforce();
    ...

### `rtpproxy_stream2uac(prompt_name, count [, set_id [, sock_pvar]])`, `rtpproxy_stream2uas(prompt_name, count [, set_id [, sock_pvar]])`

Instruct the RTPproxy to stream prompt/announcement pre-encoded with the makeann command from the RTPproxy distribution. The uac/uas suffix selects who will hear the announcement relatively to the current transaction - UAC or UAS. For example invoking the `rtpproxy_stream2uac` in the request processing block on ACK transaction will play the prompt to the UA that has generated original INVITE and ACK while `rtpproxy_stop_stream2uas` on 183 in reply processing block will play the prompt to the UA that has generated 183.

Apart from generating announcements, another possible application of this function is implementing music on hold (MOH) functionality. When count is -1, the streaming will be in loop indefinitely until the appropriate `rtpproxy_stop_stream2xxx` is issued.

In order to work correctly, functions require that the session in the RTPproxy already exists. Also those functions don't alted SDP, so that they are not substitute for calling `rtpproxy_offer` or `rtpproxy_answer`.

This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE.

Meaning of the parameters is as follows:

- prompt_name - name of the prompt to stream. Should be either absolute pathname or pathname relative to the directory where RTPproxy runs.

- count - number of times the prompt should be repeated. The value of -1 means that it will be streaming in loop indefinitely, until appropriate `rtpproxy_stop_stream2xxx` is issued.

- set_id(optional) - the set used for this call.

- sock_pvar(optional) - pvar used to store the RTPProxy socket chosen for this call.

#### `rtpproxy_stream2xxx` usage

    ...
        if (is_method("INVITE")) {
            rtpproxy_offer();
            if ($rb=~ "0\.0\.0\.0") {
                rtpproxy_stream2uas("/var/rtpproxy/prompts/music_on_hold", "-1");
            } else {
                rtpproxy_stop_stream2uas();
            };
        };
    ...

### `rtpproxy_stop_stream2uac([set_id [, sock_pvar]])`, `rtpproxy_stop_stream2uas([set_id [, sock_pvar]])`

Stop streaming of announcement/prompt/MOH started previously by the respective `rtpproxy_stream2xxx`. The uac/uas suffix selects whose announcement relatively to tha current transaction should be stopped - UAC or UAS.

Meaning of the parameters is as follows:

- set_id(optional) - the set used for this call.

- sock_pvar(optional) - pvar used to store the RTPProxy socket chosen for this call.

These functions can be used from REQUEST_ROUTE, ONREPLY_ROUTE.

### `start_recording([set_id [, sock_pvar]]) - deprecated`, `rtpproxy_start_recording([set_id [, sock_pvar]])`

This command will send a signal to the RTP-Proxy to record the RTP stream on the RTP-Proxy.

Meaning of the parameters is as follows:

- set_id(optional) - the set used for this call.

- sock_pvar(optional) - pvar used to store the RTPProxy socket chosen for this call.

This function can be used from REQUEST_ROUTE and ONREPLY_ROUTE.

#### `rtpproxy_start_recording` usage

    ...
    rtpproxy_start_recording();
    ...

## MI Commands

### `rtpproxy_enable`

Enables a rtp proxy if parameter value is greater than 0. Disables it if a zero value is given.

The first parameter is the rtp proxy url (exactly as defined in the config file).

The next parameter (optional) is the rtpproxy set ID (used for better indentification of the rtpproxy instance to be enabled, for example when a rtpproxy is used in multiple sets).

The last parameter must be a number in decimal representing the new enabled/disabled state.

NOTE: if a rtpproxy is defined multiple times (in the same or diferente sete), all its instances will be enables/disabled IF no set ID provided (as second param).

#### `rtpproxy_enable` usage

    ...
    ## disable a RTPProxy by URL only
    $ opensipsctl fifo rtpproxy_enable udp:192.168.2.133:8081 0
    ## disable a RTPProxy by URL and set ID (3)
    $ opensipsctl fifo rtpproxy_enable udp:192.168.2.133:8081 3 0
    ...

### `rtpproxy_show`

Displays all the rtp proxies and their information: set and status (disabled or not, weight and recheck_ticks).

No parameter.

#### `rtpproxy_show` usage

    ...
    $ opensipsctl fifo rtpproxy_show
    ...

### `rtpproxy_reload`

Reload rtp proxies sets from database. The function will delete all previous records and populate the list with the entries from the database table. The db_url parameter must be set if you want to use this command.

No parameter.

#### `rtpproxy_reload` usage

    ...
    $ opensipsctl fifo rtpproxy_reload
    ...

## Exported Events

### `E_RTPPROXY_STATUS`

This event is raised when a RTPProxy server changes it's status to enabled/disabled.

Parameters:

- socket - the socket that identifies the RTPProxy instance.

- status - active if the RTPProxy instance responds to probing or inactive if the instance was deactivated.