# &adminguide;

## Overview

This is a module that enables media streams to be proxied via the Sippy RTPproxy .

## Multiple RTPProxy usage

The rtpproxy module can support multiple rtpproxies for balancing/distribution and control/selection purposes.

The module allows definition of several sets of rtpproxies. Load-balancing will be performed over a set and the admin has the ability to choose what set should be used. The set is selected via its id - the id being defined 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.

The selection of the set is done from script prior using unforce_rtp_proxy(), rtpproxy_offer() or rtpproxy_answer() functions - see the set_rtp_proxy_set() function.

For backward compatibility reasons, a set with no id take by default the id 0. Also if no set is explicitly set before unforce_rtp_proxy(), rtpproxy_offer() or rtpproxy_answer() the 0 id set will be used.

IMPORTANT: if you use multiple sets, take care and use the same set for both rtpproxy_offer()/rtpproxy_answer() and unforce_rtpproxy()!!

## Dependencies

### &kamailio; Modules

The following modules must be loaded before this module:

- tm module - (optional) if you want to have rtpproxy_manage() fully functional

### External Libraries or Applications

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

- None.

## Parameters

### `rtpproxy_sock` (string)

Used to define the list of RTPProxy instances to connect to. These can be UNIX sockets or IPv4/IPv6 UDP sockets. Each modparam entry will insert sockets into a single set. If no set ID is given, the default set ID '0' will be used. To define multiple sets add the set number at the beginning of each parameter followed by '=='. Sockets can be weighted by adding '=#' to a socket where # is an integer. A socket with a weight of 2 will be chosen twice as often as one with a weight of 1.

Default value is NONE (disabled).

#### Set `rtpproxy_sock` parameter

    ...
    # single rtproxy
    modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:12221")

    # 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 disabled, the rtpproxy module 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_tout` (integer)

Timeout value in waiting for reply from RTPProxy.

Default value is 1.

#### Set `rtpproxy_tout` parameter

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

### `rtpproxy_retr` (integer)

How many times the module 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)

This parameter sets the SDP attribute used by rtpproxy to mark the message's SDP attachment with information that it have already been changed.

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")
    ...

### `timeout_socket` (string)

The parameter sets the RTP timeout socket, which is transmitted to the RTP-Proxy. It will be used by the RTP proxy to signal back that a media stream timed out.

If it is an empty string, no timeout socket will be transmitted to the RTP-Proxy.

Default value is (nothing).

#### Set `timeout_socket` parameter

    ...
    modparam("rtpproxy", "timeout_socket", "tcp:127.0.0.1:8000")
    ...

### `timeout_tag_pv` (string)

The parameter devines the AVP of the string to be provided to the RTP-Proxy.

The content of the AVP must be a valid URL-encoded string with no spaces. It will be decoded and send by the RTP proxy to the timeout socket if the media timeout has happened.

This parameter is required in order for the timeout notification mechanism to work properly.

Default value is (nothing).

#### Set `timeout_tag_pv` parameter

    ...
    modparam("rtpproxy", "timeout_tag_pv", "$avp(rtpp_ntag)")
    ...

### `ice_candidate_priority_avp` (string)

If specified and if value of the avp value is not 0, `rtpproxy_manage` function adds ICE relay candidate attributes to sdp stream(s) containing ICE candidate attributes.

If value of the avp is 1, added candidates have high priority. If value of the avp is 2 (default), added candidates have low priority.

There is no default value meaning that no ICE relay candidates are added in any circumstance.

#### Set `ice_candidate_priority_avp` parameter

    ...
    modparam("rtpproxy", "ice_candidate_priority_avp", "$avp(ice_priority)")
    ...

### `extra_id_pv` (string)

The parameter sets the PV definition to use when the b parameter is used on unforce_rtp_proxy(), rtpproxy_offer(), rtpproxy_answer() or rtpproxy_manage() command.

Default is empty, the b parameter may not be used then.

#### Set `extra_id_pv` parameter

    ...
    modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)")
    ...

### `db_url` (string)

The database URL to load rtp_proxy sets from. If this parameter is set, the module will attempt to load the rtpproxy sets from the specified database and will ignore any 'rtpproxy_sock' modparams.

Default is empty, a database will not be used.

#### Set `db_url` parameter

    ...
    modparam("rtpproxy", "db_url", "mysql://user:passwd@localhost/database")
    ...

### `table_name` (string)

The name of the table containing the rtpproxy sets.

Default value is rtpproxy.

#### Set `table_name` parameter

    ...
    modparam("rtpproxy", "table_name", "my_rtpp_sets")
    ...

### `rtp_inst_pvar` (string)

A pseudo variable to store the chosen RTPProxy address. If this parameter is set, the instance URL will be stored in the given variable.

By default, this parameter is not set.

#### Set `rtp_inst_pvar` parameter

    ...
    modparam("rtpproxy", "rtp_inst_pvar", "$avp(RTP_INSTANCE)")
    ...

#### `rtp_inst_pvar` usage

    modparam("rtpproxy", "rtpproxy_sock",
    	"udp:localhost:12221 udp:localhost:12222")
    modparam("rtpproxy", "rtp_inst_pvar", "$var(RTP_INSTANCE)")
    ...
    rtpproxy_manage("eiro");
    xlog("L_INFO", "Chose rtpp instance $var(RTP_INSTANCE)\n");
    # This will display 'udp:localhost:12222'
    ...

## Functions

### `set_rtp_proxy_set(setid)`

Sets the Id of the rtpproxy set to be used for the next unforce_rtp_proxy(), rtpproxy_offer(), rtpproxy_answer() or rtpproxy_manage() command. The parameter can be an integer or a config variable holding an integer.

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

#### `set_rtp_proxy_set` usage

    ...
    set_rtp_proxy_set("2");
    rtpproxy_offer();
    ...

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

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.

Meaning of the parameters is as follows:

- flags - flags to turn on some features.

  - 1 - append first Via branch to Call-ID when sending command to rtpproxy. This can be used to create one media session per branch on the rtpproxy. When sending a subsequent delete command to the rtpproxy, you can then stop just the session for a specific branch when passing the flag '1' or '2' in the unforce_rtpproxy, or stop all sessions for a call when not passing one of those two flags there. This is especially useful if you have serially forked call scenarios where rtpproxy gets an update command for a new branch, and then a delete command for the previous branch, which would otherwise delete the full call, breaking the subsequent lookup for the new branch. This flag is only supported by the ngcp-mediaproxy-ng rtpproxy (now named rtpengine) at the moment!

  - 2 - append second Via branch to Call-ID when sending command to rtpproxy. See flag '1' for its meaning.

  - 3 - behave like flag 1 is set for a request and like flag 2 is set for a reply.

  - a - flags that UA from which message is received doesn't support symmetric RTP. (automatically sets the 'r' flag)

  - b - append branch specific variable to Call-ID when sending command to rtpproxy. This creates one rtpproxy session per unique variable. Works similar to the 1, 2 and 3 parameter, but is useful when forking to multiple destinations on different address families or network segments, requiring different rtpproxy parameters. The variable value is taken from the extra_id_pv. When used, it must be used in every call to rtpproxy_manage(), rtpproxy_offer(), rtpproxy_answer() and rtpproxy_destroy() with the same contents of the PV. The b parameter may not be used in conjunction with the 1, 2 or 3 parameter to use the Via branch in the Call-ID.

  - l - force lookup, that is, only rewrite SDP when corresponding session already exists in the RTP proxy. By default is on when the session is to be completed.

  - i, e - these flags specify the direction of the SIP message. These flags only make sense when rtpproxy is running in bridge mode. 'i' means internal network (LAN), 'e' means external network (WAN). 'i' corresponds to rtpproxy's first interface, 'e' corresponds to rtpproxy's second interface. You always have to specify two flags to define the incoming network and the outgoing network. For example, 'ie' should be used for SIP message received from the local interface and sent out on the external interface, and 'ei' vice versa. Other options are 'ii' and 'ee'. So, for example if a SIP requests is processed with 'ie' flags, the corresponding response must be processed with 'ie' flags.

    Note: As rtpproxy in bridge mode is asymmetric per default, you have to specify the 'w' flag for clients behind NAT! See also above notes!

  - x - this flag a shortcut for using the "ie" or "ei"-flags of RTP-Proxy, in order to do automatic bridging between IPv4 on the "internal network" and IPv6 on the "external network". The distinction is done by the given IP in the SDP, e.g. an IPv4 Address will always call "ie" to the RTPProxy (IPv4(i) to IPv6(e)) and an IPv6Address will always call "ei" to the RTPProxy (IPv6(e) to IPv4(i)).

    Note: Please note, that this will only work properly with non-dual-stack user-agents or with dual-stack clients according to RFC6157 (which suggest ICE for Dual-Stack implementations). This short-cut will not work properly with RFC4091 (ANAT) compatible clients, which suggests having different m-lines with different IP-protocols grouped together.

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

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

  - 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 bandwidth overhead for low bitrate codecs, for example with G.729 going from 10ms to 100ms saves two thirds of the network bandwidth.

- ip_address - new SDP IP address.

This function can be used from ANY_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]])`

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_answer() 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.

### `rtpproxy_destroy([flags])`

Tears down the RTPProxy session for the current call.

This function can be used from ANY_ROUTE.

Meaning of the parameters is as follows:

- flags - flags to turn on some features.

  - 1 - append first Via branch to Call-ID when sending command to rtpproxy. This can be used to create one media session per branch on the rtpproxy. When sending a subsequent delete command to the rtpproxy, you can then stop just the session for a specific branch when passing the flag '1' or '2' in the unforce_rtpproxy, or stop all sessions for a call when not passing one of those two flags there. This is especially useful if you have serially forked call scenarios where rtpproxy gets an update command for a new branch, and then a delete command for the previous branch, which would otherwise delete the full call, breaking the subsequent lookup for the new branch. This flag is only supported by the ngcp-mediaproxy-ng rtpproxy (now named rtpengine) at the moment!

  - 2 - append second Via branch to Call-ID when sending command to rtpproxy. See flag '1' for its meaning.

  - b - append branch specific variable to Call-ID when sending command to rtpproxy. See rtpproxy_offer() for details.

  - t - do not include To tag to delete command to rtpproxy thus causing full call to be deleted. Useful for deleting unused rtpproxy call when 200 OK is received on a branch, where rtpproxy is not needed.

#### `rtpproxy_destroy` usage

    ...
    rtpproxy_destroy();
    ...

### `unforce_rtp_proxy()`

Same as rtpproxy_destroy().

### `rtpproxy_manage([flags [, ip_address]])`

Manage the RTPProxy session - it combines the functionality of rtpproxy_offer(), rtpproxy_answer() and unforce_rtpproxy(), detecting internally based on message type and method which one to execute.

It can take the same parameters as `rtpproxy_offer()`. The flags parameter to rtpproxy_manage() can be a configuration variable containing the flags as a string.

Functionality:

- If INVITE with SDP, then do `rtpproxy_offer()`

- If INVITE with SDP, when the tm module is loaded, mark transaction with internal flag FL_SDP_BODY to know that the 1xx and 2xx are for `rtpproxy_answer()`

- If ACK with SDP, then do `rtpproxy_answer()`

- If BYE or CANCEL, or called within a FAILURE_ROUTE[], then call `unforce_rtpproxy()`. Be careful with calling this function after resuming a suspended transaction (e.g., after t_continue()), because the context of executed route is FAILURE ROUTE (in other words, rtpproxy_manage() in the route block of t_continue() does the same as in failure_route; use a branch route to engage rtpengine for a forwarded branch after resuming the transaction).

- It does nothing if used inside event_route[tm:branch-failure:...]

- If reply to INVITE with code >= 300 do `unforce_rtpproxy()`

- If reply with SDP to INVITE having code 1xx and 2xx, then do `rtpproxy_answer()` if the request had SDP or tm is not loaded, otherwise do `rtpproxy_offer()`

This function can be used from ANY_ROUTE.

#### `rtpproxy_manage` usage

    ...
    rtpproxy_manage();
    ...

### `rtpproxy_stream2uac(prompt_name, count)`,

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, these functions require that a session in the RTPproxy already exists. Also those functions don't alter the SDP, so that they are not a 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. A value of -1 means that it will be streaming in a loop indefinitely, until the appropriate `rtpproxy_stop_stream2xxx` is issued.

#### `rtpproxy_stream2xxx` usage

    ...
        if (is_method("INVITE")) {
            rtpproxy_offer();
            if (is_audio_on_hold()) {
                rtpproxy_stream2uas("/var/rtpproxy/prompts/music_on_hold", "-1");
            } else {
                rtpproxy_stop_stream2uas();
            };
        };
    ...

### `rtpproxy_stream2uas(prompt_name, count)`

See function `rtpproxy_stream2uac(prompt_name, count)`.

### `rtpproxy_stop_stream2uac()`,

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.

These functions can be used from REQUEST_ROUTE, ONREPLY_ROUTE.

### `rtpproxy_stop_stream2uas()`

See function `rtpproxy_stop_stream2uac()`.

#### `rtpproxy_stop_stream2uas` usage

    ...
        if (is_method("INVITE")) {
            rtpproxy_offer();
            if (is_audio_on_hold()) {
                rtpproxy_stream2uas("/var/rtpproxy/prompts/music_on_hold", "-1");
            } else {
                rtpproxy_stop_stream2uas();
            };
        };
    ...

### `start_recording()`

This function will send a signal to the RTP-Proxy to record the RTP stream on the RTP-Proxy. This function is only supported by Sippy RTPproxy at the moment!

This function can be used from REQUEST_ROUTE and ONREPLY_ROUTE.

#### `start_recording` usage

    ...
    start_recording();
    ...

## Exported Variables

### `$rtppstat`

Returns the RTP-Statistics from the RTP-Proxy. The RTP-Statistics from the RTP-Proxy are provided as a string and it does contain several packet-counters. The statistics must be retrieved before the session is deleted (before `unforce_rtpproxy()`). It is the output of RTPProxy 'Q' command.

#### $rtppstat Usage

    ...
        append_hf("X-RTP-Statistics: $rtppstat\r\n");
    ...

### `$rtpstat`

Same as $rtppstat.

## RPC 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 second parameter value must be a number in decimal.

NOTE: if a rtpproxy is defined multiple times (in the same or different sets), all of its instances will be enabled/disabled.

#### `rtpproxy.enable` usage

    ...
    $ &kamctl; rpc rtpproxy.enable udp:192.168.2.133:8081 0
    ...

### `rtpproxy.list`

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

No parameter.

#### `rtpproxy.list` usage

    ...
    $ &kamctl; rpc rtpproxy.list
    ...