rtpproxy Module
     __________________________________________________________

   Table of Contents

   1. Admin Guide

        1.1. Overview
        1.2. Multiple RTPProxy usage
        1.3. RTPProxy timeout notifications
        1.4. Dependencies

              1.4.1. OpenSIPS Modules
              1.4.2. External Libraries or Applications

        1.5. Exported Parameters

              1.5.1. rtpproxy_sock (string)
              1.5.2. rtpproxy_disable_tout (integer)
              1.5.3. rtpproxy_timeout (string)
              1.5.4. rtpproxy_autobridge (integer)
              1.5.5. rtpproxy_tout (integer)
              1.5.6. rtpproxy_retr (integer)
              1.5.7. default_set (integer)
              1.5.8. nortpproxy_str (string)
              1.5.9. db_url (string)
              1.5.10. db_table (string)
              1.5.11. rtpp_socket_col (string)
              1.5.12. set_id_col (string)
              1.5.13. rtpp_notify_socket (string)

        1.6. Exported Functions

              1.6.1. rtpproxy_engage([[flags][, [ip_address][,
                      [set_id][, [sock_var][, ret_var]]]]])

              1.6.2. rtpproxy_offer([[flags][, [ip_address][,
                      [set_id][, [sock_var][, ret_var]]]])

              1.6.3. rtpproxy_answer([[flags][, [ip_address][,
                      [set_id][, [sock_var][, ret_var]]]]])

              1.6.4. rtpproxy_unforce([[set_id][, sock_var]])
              1.6.5. rtpproxy_stream2uac(prompt_name, count[,
                      [set_id][, sock_var]]),
                      rtpproxy_stream2uas(prompt_name, count[,
                      [set_id][, sock_var]])

              1.6.6. rtpproxy_stop_stream2uac([[set_id][,
                      sock_var]]),
                      rtpproxy_stop_stream2uas([[set_id][,
                      sock_var]])

              1.6.7. rtpproxy_start_recording([[set_id][,
                      [sock_var][, [flags][, [destination][,
                      mediastream]]]]])

              1.6.8. rtpproxy_stats(up_pvar, down_var, sent_var,
                      fail_var[, [set_id][, sock_var]])

              1.6.9. rtpproxy_all_stats(stats_avp[, [set_id][,
                      sock_var]])

        1.7. Exported MI Functions

              1.7.1. rtpproxy_enable
              1.7.2. rtpproxy_show
              1.7.3. rtpproxy_reload

        1.8. Exported Events

              1.8.1. E_RTPPROXY_STATUS
              1.8.2. E_RTPPROXY_DTMF

   2. Frequently Asked Questions
   3. Contributors

        3.1. By Commit Statistics
        3.2. By Commit Activity

   4. Documentation

        4.1. Contributors

   List of Tables

   3.1. Top contributors by DevScore^(1), authored commits^(2) and
          lines added/removed^(3)

   3.2. Most recently active contributors^(1) to this module

   List of Examples

   1.1. Set rtpproxy_sock parameter
   1.2. Set rtpproxy_disable_tout parameter
   1.3. Set rtpproxy_timeout parameter to 200ms
   1.4. Enable auto-bridging feature
   1.5. Set rtpproxy_retr parameter
   1.6. Set default_set parameter
   1.7. Set nortpproxy_str parameter
   1.8. Set db_url parameter
   1.9. Set db_table parameter
   1.10. Set rtpp_socket_col parameter
   1.11. Set set_id parameter
   1.12. Set rtpp_notify_socket parameter
   1.13. rtpproxy_engage usage
   1.14. rtpproxy_offer usage
   1.15. rtpproxy_answer usage
   1.16. rtpproxy_unforce usage
   1.17. rtpproxy_stream2xxx usage
   1.18. rtpproxy_start_recording usage
   1.19. rtpproxy_stats usage
   1.20. rtpproxy_all_stats usage
   1.21. rtpproxy_enable usage
   1.22. rtpproxy_show usage
   1.23. rtpproxy_reload usage

Chapter 1. Admin Guide

1.1. Overview

   This module is used by OpenSIPS 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 UAc or
   UAs.

1.2. 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 OpenSIPS 2.1, engage_rtp_proxy(),
   unforce_rtp_proxy() and start_recording() functions have been
   fully replaced by rtpproxy_engage(), rtpproxy_unforce() and
   rtpproxy_start_recording().

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

1.3. 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.

   Note that RTPProxy version v2.0.0 has integrated this feature
   upstream, therefore this patch is no longer needed.

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

   Start OpenSIPS 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” OpenSIPS 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:
   https://opensips.org/pub/rtpproxy/

1.4. Dependencies

1.4.1. OpenSIPS 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.

1.4.2. External Libraries or Applications

   The following libraries or applications must be installed
   before running OpenSIPS with this module loaded:
     * None.

1.5. Exported Parameters

1.5.1. 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.

   The definition also supports to specify a different IP that
   will be advertised instead of the one returned by RTPProxy.
   This is useful when having multiple RTPProxy servers that are
   located behind NAT, and listen only on private intefaces, but
   need to advertise a public one.

   Default value is “NONE” (disabled).

   Example 1.1. 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")
modparam("rtpproxy", "rtpproxy_sock",
        "2 == udp:localhost:12225|8.8.8.8")
...

1.5.2. 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”.

   Example 1.2. Set rtpproxy_disable_tout parameter
...
modparam("rtpproxy", "rtpproxy_disable_tout", 20)
...

1.5.3. rtpproxy_timeout (string)

   Timeout value in waiting for reply from RTPProxy.

   Default value is “1”.

   Example 1.3. Set rtpproxy_timeout parameter to 200ms
...
modparam("rtpproxy", "rtpproxy_timeout", "0.2")
...

1.5.4. rtpproxy_autobridge (integer)

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

   Default value is “0”.

   Example 1.4. Enable auto-bridging feature
...
modparam("rtpproxy", "rtpproxy_autobridge", 1)
...

1.5.5. rtpproxy_tout (integer)

   Obsolete. see rtpproxy_timeout.

1.5.6. rtpproxy_retr (integer)

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

   Default value is “5”.

   Example 1.5. Set rtpproxy_retr parameter
...
modparam("rtpproxy", "rtpproxy_retr", 2)
...

1.5.7. default_set (integer)

   The parameter indicates the default RTPProxy set to be used
   when provisioning an engine in the config file without an
   explicit set, or when calling one of the rtpproxy_*() functions
   without an explicit set.

   Default value is set “0”.

   Example 1.6. Set default_set parameter
...
modparam("rtpproxy", "default_set", 1)
...

1.5.8. 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.

Note

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

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

   Example 1.7. Set nortpproxy_str parameter
...
modparam("rtpproxy", "nortpproxy_str", "a=sdpmangled:yes\r\n")
...

1.5.9. 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”.

   Example 1.8. Set db_url parameter
...
modparam("rtpproxy", "db_url",
                "mysql://opensips:opensipsrw@192.168.2.132/opensips")
...


1.5.10. 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”.

   Example 1.9. Set db_table parameter
...
modparam("rtpproxy", "db_table", "nh_sockets")
...


1.5.11. rtpp_socket_col (string)

   The name rtpp socket column in the database table.

   Default value is “rtpproxy_sock”.

   Example 1.10. Set rtpp_socket_col parameter
...
modparam("rtpproxy", "rtpp_socket_col", "rtpp_socket")
...


1.5.12. set_id_col (string)

   The name set id column in the database table.

   Default value is “set_id”.

   Example 1.11. Set set_id parameter
...
modparam("rtpproxy", "set_id_col", "rtpp_set_id")
...


1.5.13. rtpp_notify_socket (string)

   The socket OpenSIPS listens for notifications from RTPProxy.
   Currently OpenSIPS can receive RTP timeout and DTMF events.

   Default value is “NULL” - no notifications are received.

   Example 1.12. Set rtpp_notify_socket parameter
...
modparam("rtpproxy", "rtpp_notify_socket", "tcp:10.10.10.10:9999")

# use an UNIX socket
modparam("rtpproxy", "rtpp_notify_socket", "unix:/tmp/rtpproxy.unix")
# or
modparam("rtpproxy", "rtpp_notify_socket", "/tmp/rtpproxy.unix")
...


1.6. Exported Functions

1.6.1.  rtpproxy_engage([[flags][, [ip_address][, [set_id][,
[sock_var][, ret_var]]]]])

   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 OpenSIPS is not aware of the RTPProxy
   configuration), so you might face an undefined behavior.

   Meaning of the parameters is as follows:
     * flags(string, 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).
          + k - only create RTPProxy session, but do not modify
            the SDP body. This is useful when you only want to
            inject some media, but do not want to engage RTPProxy
            in the entire call.
          + 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.
            NOTE: when using RTPProxy in bridge mode, all sessions
            are considered asymmetric (as oposed to symmetric if
            used in normal mode). If you have symmetric clients
            (this is the most common scenario), you'll have to
            force the s!
          + 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[<SOCKET>] - flags that enables the notification
            timeout for the session. One can specify an optional
            "advertised" socket between the < and > tags. If the
            socket is not specified, the value of
            rtpp_notify_socket is used.
          + d[NNN] - enables DTMF notifications for this call. One
            can optionally specify the payload type that DTMF will
            be used for this call - it it is not specified,
            RTPProxy uses the 101 pt. NOTE: this feature is
            currently only available in the RTPProxy rtpp_2_1_dtmf
            branch.
          + tNN - can be used to specify a RTP ttl for the caller.
            The NN represents the timeout in seconds for that
            stream. This can be useful in music on hold scenarios
            where only one client is sending RTP.
          + TNN - Similar to the tNN paramaeter, but used for
            tuning the calllee's ttl for RTP.
          + 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(string, optional) - new SDP IP address.
     * set_id(int, optional) - the set used for this call.
     * sock_var(var, optional) - variable used to store the
       RTPProxy socket chosen for this call. Note that the
       variable will only be populated in the initial request.
     * ret_var(var, optional) - variable used to print the IP and
       port the RTPProxy server is using for this call. This is
       useful especially when using the rtp_cluster, which can
       advertise multiple servers behind it. The format of the
       value returned is IP:port. Note that the variable will only
       be populated in the initial request.

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

   Example 1.13. 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");
        }
}
...

1.6.2.  rtpproxy_offer([[flags][, [ip_address][, [set_id][,
[sock_var][, ret_var]]]])

   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.

   Example 1.14. 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();
...
}

1.6.3.  rtpproxy_answer([[flags][, [ip_address][, [set_id][,
[sock_var][, ret_var]]]]])

   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.

   Example 1.15. rtpproxy_answer usage

   See rtpproxy_offer() function example above for example.

1.6.4.  rtpproxy_unforce([[set_id][, sock_var]])

   Tears down the RTPProxy session for the current call.

   Meaning of the parameters is as follows:
     * set_id(int, optional) - the set used for this call.
     * sock_var(var, optional) - variable used to store the
       RTPProxy socket chosen for this call.

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

   Example 1.16. rtpproxy_unforce usage
...
rtpproxy_unforce();
...

1.6.5.  rtpproxy_stream2uac(prompt_name, count[, [set_id][,
sock_var]]), rtpproxy_stream2uas(prompt_name, count[, [set_id][,
sock_var]])

   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 (string) - name of the prompt to stream. Should
       be either absolute pathname or pathname relative to the
       directory where RTPproxy runs.
     * count (int) - 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(int, optional) - the set used for this call.
     * sock_var(var, optional) - variable used to store the
       RTPProxy socket chosen for this call.

   Example 1.17. 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();
        };
    };
...

1.6.6.  rtpproxy_stop_stream2uac([[set_id][, sock_var]]),
rtpproxy_stop_stream2uas([[set_id][, sock_var]])

   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(int, optional) - the set used for this call.
     * sock_var(var, optional) - variable used to store the
       RTPProxy socket chosen for this call.

   These functions can be used from REQUEST_ROUTE, ONREPLY_ROUTE.

1.6.7.  rtpproxy_start_recording([[set_id][, [sock_var][, [flags][,
[destination][, mediastream]]]]])

   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(int, optional) - the set used for this call.
     * sock_var(var, optional) - variable used to store the
       RTPProxy socket chosen for this call.
     * flags(string, optional) - a list of flags passed to
       RTPProxy for the recording. Currently only s is supported,
       and it indicates that RTPProxy should record both audio
       legs in a single file. Note that this feature is available
       starting with RTPProxy 2.0.
     * destination(string, optional) - the destination of the
       recording. If it has the udp:IP:port format, RTPProxy sends
       the RTP stream to that IP:port remote destination.
       Otherwise, destination represents the name of the file in
       the recording directory.
     * mediastream(int, optional) - this parameter is only used if
       the destination is specified, and represents the index of
       media stream to record/copy, starting from 1. If this
       parameter is missing, OpenSIPS instructs RTPProxy to copy
       all the streams.

   This function can be used from REQUEST_ROUTE and ONREPLY_ROUTE.

   Example 1.18. rtpproxy_start_recording usage
...
rtpproxy_start_recording();

# copy RTP stream to a different listener
rtpproxy_start_recording(,,,"udp:127.0.0.1:60000");

# copy only first RTP stream (audio stream)
rtpproxy_start_recording(,,,"udp:127.0.0.1:60000", 1);
...

1.6.8.  rtpproxy_stats(up_pvar, down_var, sent_var, fail_var[,
[set_id][, sock_var]])

   This command gathers call RTP statistics from RTP-Proxy.

   Meaning of the parameters is as follows:
     * up_var (var) - the variable used to return the packets sent
       by upstream for this call.
     * down_var (var) - the variable used to return the packets
       sent by downstream for this call.
     * sent_var (var) - the variable used to return the total
       number of packets sent for this call.
     * up_var (var) - the variable used to return the number of
       failed packets for this call.
     * set_id(int, optional) - the set used for this call.
     * sock_var(var, optional) - variable used to store the
       RTPProxy socket chosen for this call.

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

   Example 1.19. rtpproxy_stats usage
...
rtpproxy_stats($var(up),$var(down),$var(sent),$var(fail));
xlog("RTP statistics for $ci: up=$var(up) down=$var(down) sent=$var(sent
) fail=$var(fail)\n");
...

1.6.9.  rtpproxy_all_stats(stats_avp[, [set_id][, sock_var]])

   This command gathers all RTP statistics available from
   RTP-Proxy. All the returned values stored in an AVP that can be
   further read by indexing the AVP.

   This command is only available starting with RTPProxy 2.1
   realease.

   Meaning of the parameters is as follows:
     * stats_avp (var) - an AVP where the statistics will be
       stored. This AVP can be further indexed to get a specific
       statistic.
     * set_id(int, optional) - the set used for this call.
     * sock_var(var, optional) - variable used to store the
       RTPProxy socket chosen for this call.

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

   Each statistic is stored at a specific index as it follows:
     * ttl - $avp(ret) / $(avp(ret)[0])
     * pkts_ia - $(avp(ret)[1])
     * pkts_io - $(avp(ret)[2])
     * relayed - $(avp(ret)[3])
     * dropped - $(avp(ret)[4])
     * rtpa_set - $(avp(ret)[5])
     * rtpa_rcvd - $(avp(ret)[6])
     * rtpa_dups - $(avp(ret)[7])
     * rtpa_lost - $(avp(ret)[8])
     * rtpa_perrs - $(avp(ret)[9])

   Example 1.20. rtpproxy_all_stats usage
...
rtpproxy_all_stats($avp(stats));
xlog("RTP statistics for $ci: dropped=$(avp(stats)[4])\n");
...

1.7. Exported MI Functions

1.7.1. rtpproxy_enable

   Enables/Disables a rtp proxy.

   Parameters:
     * url - the rtp proxy url (exactly as defined in the config
       file).
     * setid (optional) - 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).
     * enable - 1 - enable, 0 - disable. the config file).

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

   Example 1.21.  rtpproxy_enable usage
...
## disable a RTPProxy by URL only
$ opensips-cli -x mi rtpproxy_enable udp:192.168.2.133:8081 0
## disable a RTPProxy by URL and set ID (3)
$ opensips-cli -x mi rtpproxy_enable udp:192.168.2.133:8081 3 0
...

1.7.2. rtpproxy_show

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

   No parameter.

   Example 1.22.  rtpproxy_show usage
...
$ opensips-cli -x mi rtpproxy_show
...

1.7.3. 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.

   Example 1.23.  rtpproxy_reload usage
...
$ opensips-cli -x mi rtpproxy_reload
...

1.8. Exported Events

1.8.1.  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.

1.8.2.  E_RTPPROXY_DTMF

   This event is raised when a RTPProxy server sends a DTMF
   notification to OpenSIPS. In order to catch RFC 2833/4733 DTMF
   events, you need to provide the d flag to rtpproxy_offer()/
   rtpproxy_answer().

   Parameters:
     * digit - the digit pressed.
     * duration - the duration of the event.
     * volume - the volume of the event.
     * id - represents the identifier of the call for which that
       event was received.
     * is_callid - is 0 if the id parameter represents the Dialog
       ID, or 1 if it is a callid.
     * stream - indicates the stream index of the RTPProxy
       session. It is normally 0 if the caller sent the DTMF, or 1
       if the callee sent it.

Chapter 2. Frequently Asked Questions

   2.1.

       What happened with “rtpproxy_disable” parameter?

       It was removed as it became obsolete - now “rtpproxy_sock” can
       take empty value to disable the rtpproxy functionality.

   2.2.

       Where can I find more about OpenSIPS?

       Take a look at https://opensips.org/.

   2.3.

       Where can I post a question about this module?

       First at all check if your question was already answered on one
       of our mailing lists:
         * User Mailing List -
           http://lists.opensips.org/cgi-bin/mailman/listinfo/users
         * Developer Mailing List -
           http://lists.opensips.org/cgi-bin/mailman/listinfo/devel

       E-mails regarding any stable OpenSIPS release should be sent to
       <users@lists.opensips.org> and e-mails regarding development
       versions should be sent to <devel@lists.opensips.org>.

       If you want to keep the mail private, send it to
       <users@lists.opensips.org>.

   2.4.

       How can I report a bug?

       Please follow the guidelines provided at:
       https://github.com/OpenSIPS/opensips/issues.

Chapter 3. Contributors

3.1. By Commit Statistics

   Table 3.1. Top contributors by DevScore^(1), authored
   commits^(2) and lines added/removed^(3)
     Name DevScore Commits Lines ++ Lines --
   1. Razvan Crainea (@razvancrainea) 176 117 3777 1657
   2. Maksym Sobolyev (@sobomax) 41 3 4527 3
   3. Bogdan-Andrei Iancu (@bogdan-iancu) 27 23 123 116
   4. Vlad Patrascu (@rvlad-patrascu) 23 8 362 699
   5. Liviu Chircu (@liviuchircu) 20 16 91 183
   6. Ovidiu Sas (@ovidiusas) 7 5 31 22
   7. Vlad Paiu (@vladpaiu) 6 4 15 9
   8. Peter Lemenkov (@lemenkov) 5 3 60 9
   9. Ryan Bullock (@rrb3942) 4 2 68 9
   10. robdyck 4 2 8 4

   All remaining contributors: Dave Sidwell (@davesidwell),
   Ezequiel Lovelle, Christophe Sollet (@csollet), Anca Vamanu,
   Mikko Lehto, Dan Pascu (@danpascu), Walter Doekes (@wdoekes),
   Dusan Klinec, Julián Moreno Patiño, Zero King.

   (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

3.2. By Commit Activity

   Table 3.2. Most recently active contributors^(1) to this module
                      Name                   Commit Activity
   1.  Ovidiu Sas (@ovidiusas)             Mar 2011 - Jun 2020
   2.  Razvan Crainea (@razvancrainea)     Mar 2011 - May 2020
   3.  Bogdan-Andrei Iancu (@bogdan-iancu) Mar 2011 - Apr 2020
   4.  robdyck                             Apr 2020 - Apr 2020
   5.  Zero King                           Mar 2020 - Mar 2020
   6.  Vlad Patrascu (@rvlad-patrascu)     May 2017 - Nov 2019
   7.  Dan Pascu (@danpascu)               May 2019 - May 2019
   8.  Ryan Bullock (@rrb3942)             Mar 2013 - Jan 2019
   9.  Liviu Chircu (@liviuchircu)         Jul 2012 - Nov 2018
   10. Peter Lemenkov (@lemenkov)          Dec 2011 - Jun 2018

   All remaining contributors: Maksym Sobolyev (@sobomax), Julián
   Moreno Patiño, Vlad Paiu (@vladpaiu), Dusan Klinec, Dave
   Sidwell (@davesidwell), Ezequiel Lovelle, Mikko Lehto, Walter
   Doekes (@wdoekes), Christophe Sollet (@csollet), Anca Vamanu.

   (1) including any documentation-related commits, excluding
   merge commits

Chapter 4. Documentation

4.1. Contributors

   Last edited by: Razvan Crainea (@razvancrainea), Zero King,
   Vlad Patrascu (@rvlad-patrascu), Liviu Chircu (@liviuchircu),
   Peter Lemenkov (@lemenkov), Julián Moreno Patiño, Bogdan-Andrei
   Iancu (@bogdan-iancu), Mikko Lehto, Ryan Bullock (@rrb3942),
   Maksym Sobolyev (@sobomax), Ovidiu Sas (@ovidiusas).

   Documentation Copyrights:

   Copyright © 2005 Voice Sistem SRL

   Copyright © 2003-2008 Sippy Software, Inc.
