rtpengine Module
     __________________________________________________________

   Table of Contents

   1. Admin Guide

        1.1. Overview
        1.2. Multiple RTP proxy usage
        1.3. Dependencies

              1.3.1. OpenSIPS Modules
              1.3.2. External Libraries or Applications

        1.4. Exported Parameters

              1.4.1. rtpengine_sock (string)
              1.4.2. rtpengine_disable_tout (integer)
              1.4.3. rtpengine_tout (integer)
              1.4.4. rtpengine_retr (integer)
              1.4.5. rtpengine_timer_interval (integer)
              1.4.6. notification_sock (string)
              1.4.7. extra_id_pv (string)
              1.4.8. setid_avp (string)
              1.4.9. error_pv (string)
              1.4.10. db_url (string)
              1.4.11. db_table (string)
              1.4.12. socket_column (string)
              1.4.13. set_column (string)

        1.5. Exported Functions

              1.5.1. rtpengine_use_set(setid)
              1.5.2. rtpengine_offer([flags[, sock_var[,
                      sdp_pvar[, body]]]])

              1.5.3. rtpengine_answer([flags[, sock_pvar[,
                      sdp_pvar[, body]]]])

              1.5.4. rtpengine_delete([flags[, sock_var]])
              1.5.5. rtpengine_manage([flags[, sock_var[,
                      sdp_var[, body]]]])

              1.5.6. rtpengine_start_recording([flags [,
                      sock_var]])

              1.5.7. rtpengine_stop_recording([flags [,
                      sock_var]])

              1.5.8. rtpengine_play_media(flags, [duration_spec[,
                      sock_var[, sockvar]]])

              1.5.9. rtpengine_stop_media([flags[, sockvar]])
              1.5.10. rtpengine_block_media([flags[, sockvar]])
              1.5.11. rtpengine_unblock_media([flags[, sockvar]])
              1.5.12. rtpengine_block_dtmf([flags[, sockvar]])
              1.5.13. rtpengine_unblock_dtmf([flags[, sockvar]])
              1.5.14. rtpengine_start_forwarding([flags[,
                      sockvar]])

              1.5.15. rtpengine_stop_forwarding([flags[,
                      sockvar]])

              1.5.16. rtpengine_play_dtmf(code, [flags[,
                      sockvar]])

        1.6. Exported Pseudo-Variables

              1.6.1. $rtpstat
              1.6.2. $rtpstat(STAT)[index]
              1.6.3. $rtpquery

        1.7. Exported MI Functions

              1.7.1. rtpengine_enable
              1.7.2. rtpengine_show
              1.7.3. rtpengine_reload
              1.7.4. teardown

        1.8. Exported Events

              1.8.1. E_RTPENGINE_NOTIFICATION
              1.8.2. E_RTPENGINE_STATUS

   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 rtpengine_sock parameter
   1.2. Set rtpengine_disable_tout parameter
   1.3. Set rtpengine_tout parameter
   1.4. Set rtpengine_retr parameter
   1.5. Set rtpengine_timer_interval parameter
   1.6. Set notification_sock parameter
   1.7. Set extra_id_pv parameter
   1.8. Set setid_avp parameter
   1.9. Set error_pv parameter
   1.10. Set db_url parameter
   1.11. Set db_table parameter
   1.12. Set socket_column parameter
   1.13. Set set_column parameter
   1.14. rtpengine_use_set usage
   1.15. rtpengine_offer usage
   1.16. rtpengine_offer usage with body replace
   1.17. rtpengine_offer usage with call recording
   1.18. rtpengine_offer usage for transcoding
   1.19. rtpengine_answer usage
   1.20. rtpengine_delete usage
   1.21. rtpengine_manage usage
   1.22. rtpengine_start_recording usage
   1.23. rtpengine_stop_recording usage
   1.24. Ringback tone using rtpengine_play_media
   1.25. Manage music on hold using rtpengine_play_media
   1.26. Ringback tone stop using rtpengine_stop_media
   1.27. Example of rtpengine_block_media usage
   1.28. Example of rtpengine_unblock_media usage
   1.29. Example of rtpengine_block_dtmf usage
   1.30. Example of rtpengine_unblock_dtmf usage
   1.31. Example of rtpengine_start_forwarding usage
   1.32. Example of rtpengine_stop_forwarding usage
   1.33. Example of rtpengine_play_dtmf usage
   1.34. $rtpstat Usage
   1.35. $rtpstat(STAT)
   1.36. $rtpquery Usage
   1.37. rtpengine_enable usage
   1.38. rtpengine_show usage
   1.39. rtpengine_reload usage
   1.40. teardown usage

Chapter 1. Admin Guide

1.1. Overview

   This is a module that enables media streams to be proxied via
   an RTP proxy. The only RTP proxy currently known to work with
   this module is the Sipwise rtpengine
   https://github.com/sipwise/rtpengine. The rtpengine module is a
   modified version of the original rtpproxy module using a new
   control protocol. The module is designed to be a drop-in
   replacement for the old module from a configuration file point
   of view, however due to the incompatible control protocol, it
   only works with RTP proxies which specifically support it.

1.2. Multiple RTP proxy usage

   The rtpengine module can support multiple RTP proxies for
   balancing/distribution and control/selection purposes.

   The module allows definition of several sets of rtpengines.
   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 “rtpengine_sock” module parameter definition for syntax
   description.

   The balancing inside a set is done automatically by the module
   based on the weight of each RTP proxy from the set.

   The selection of the set is done from script prior using
   rtpengine_delete(), rtpengine_offer() or rtpengine_answer()
   functions - see the rtpengine_use_set() function.

   Another way to select the set is to define setid_avp module
   parameter and assign setid to the defined avp before calling
   rtpengine_offer() or rtpengine_manage() function. If forwarding
   of the requests fails and there is another branch to try,
   remember to unset the avp after calling rtpengine_delete()
   function.

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

   IMPORTANT: if you use multiple sets, take care and use the same
   set for both rtpengine_offer()/rtpengine_answer() and
   rtpengine_delete()!! If the set was selected using setid_avp,
   the avp needs to be set only once before rtpengine_offer() or
   rtpengine_manage() call.

1.3. Dependencies

1.3.1. OpenSIPS Modules

   The following modules must be loaded before this module:
     * tm module - (optional) if you want to have
       rtpengine_manage() fully functional

1.3.2. External Libraries or Applications

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

1.4. Exported Parameters

1.4.1. rtpengine_sock (string)

   Definition of socket(s) used to connect to (a set) RTP proxy.
   It may specify a UNIX socket or an IPv4/IPv6 UDP socket. If the
   protocol part (i.e. “udp:”) is missing, the socket is treated
   as a UNIX socket.

   Default value is “NONE” (disabled).

   Example 1.1. Set rtpengine_sock parameter
...
# single rtproxy
modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221")
# multiple rtproxies for LB
modparam("rtpengine", "rtpengine_sock",
        "udp:localhost:12221 udp:localhost:12222")
# multiple sets of multiple rtproxies
modparam("rtpengine", "rtpengine_sock",
        "1 == udp:localhost:12221 udp:localhost:12222")
modparam("rtpengine", "rtpengine_sock",
        "2 == udp:localhost:12225")
...

1.4.2. rtpengine_disable_tout (integer)

   Once an RTP proxy was found unreachable and marked as disabled,
   the rtpengine module will not attempt to establish
   communication to that RTP proxy for rtpengine_disable_tout
   seconds.

   Default value is “60”.

   Example 1.2. Set rtpengine_disable_tout parameter
...
modparam("rtpengine", "rtpengine_disable_tout", 20)
...

1.4.3. rtpengine_tout (integer)

   Timeout value in waiting for reply from RTP proxy.

   Default value is “1”.

   Example 1.3. Set rtpengine_tout parameter
...
modparam("rtpengine", "rtpengine_tout", 2)
...

1.4.4. rtpengine_retr (integer)

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

   Default value is “5”.

   Example 1.4. Set rtpengine_retr parameter
...
modparam("rtpengine", "rtpengine_retr", 2)
...

1.4.5. rtpengine_timer_interval (integer)

   Frequency to scan rtpengine sets for disabled node probing.
   Probing is done outside the SIP processing context and in a
   separate timer routine. Disabled nodes are probed for
   re-enablement after rtpengine_disable_tout seconds. Setting
   this value too high can lead to unexpectedly large disabled
   interval as the max interval before probing is
   (rtpengine_timer_interval + rtpengine_disable_tout) seconds.

   Default value is “5”.

   Example 1.5. Set rtpengine_timer_interval parameter
...
modparam("rtpengine", "rtpengine_timer_interval", 1)
...

1.4.6. notification_sock (string)

   An UDP socket formatted as IP:port that indicates the listening
   IP and port OpenSIPS will bind for to receive notifications
   (such as DTMF events) from RTPengine.

   Every notification received from RTPengine will trigger an
   E_RTPENGINE_NOTIFICATION event.

   Default value is “none” - notifications are ignored.

   Example 1.6. Set notification_sock parameter
...
modparam("rtpengine", "notification_sock", "127.0.0.1:9999")
...

1.4.7. extra_id_pv (string)

   The parameter sets the PV definition to use when the
   “via-branch=extra” option is used on the rtpengine_delete(),
   rtpengine_offer(), rtpengine_answer() or rtpengine_manage()
   commands.

   Default is empty, the “via-branch=extra” option may not be used
   then.

   Example 1.7. Set extra_id_pv parameter
...
modparam("rtpengine", "extra_id_pv", "$avp(extra_id)")
...

1.4.8. setid_avp (string)

   The parameter defines an AVP that, if set, determines which RTP
   proxy set rtpengine_offer(), rtpengine_answer(),
   rtpengine_delete(), and rtpengine_manage() functions use.

   There is no default value.

   Example 1.8. Set setid_avp parameter
...
modparam("rtpengine", "setid_avp", "$avp(setid)")
...

1.4.9. error_pv (string)

   The parameter defines a variable that shall be populated by RTP
   when one of the rtpengine_* functions fail.

   There is no default value.

   Example 1.9. Set error_pv parameter
...
modparam("rtpengine", "error_pv", "$var(rtpengine_error)")
...

1.4.10. db_url (string)

   Database URL, used to load RTPEngines sockets from db, instead
   of specifying them in the script (rtpengine_sock module
   parameter).

   Default value is “NULL”, no database is used.

   Example 1.10. Set db_url parameter
...
modparam("rtpengine", "db_url",
                "mysql://opensips:opensipsrw@localhost/opensips")
...

1.4.11. db_table (string)

   The table where the RTPEngines sockets are stored. Used when
   Database URL is provisioned.

   Default value is “rtpengine”.

   Example 1.11. Set db_table parameter
...
modparam("rtpengine", "db_table", "rtpengine_new")
...

1.4.12. socket_column (string)

   The name of the rtpengine socket column in the database table.

   Default value is “socket”.

   Example 1.12. Set socket_column parameter
...
modparam("rtpengine", "socket_column", "sock")
...

1.4.13. set_column (string)

   The name of the rtpengine set column in the database table.

   Default value is “set_id”.

   Example 1.13. Set set_column parameter
...
modparam("rtpengine", "set_column", "set_new")
...

1.5. Exported Functions

1.5.1.  rtpengine_use_set(setid)

   Sets the ID of the RTP proxy set to be used for the next
   rtpengine_delete(), rtpengine_offer(), rtpengine_answer() or
   rtpengine_manage() command. The parameter is an integer.

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

   Example 1.14. rtpengine_use_set usage
...
rtpengine_use_set(2);
rtpengine_offer();
...

1.5.2.  rtpengine_offer([flags[, sock_var[, sdp_pvar[, body]]]])

   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(string, optional) - flags to turn on some features.
       The “flags” string is a list of space-separated items. Each
       item is either an individual token, or a token in
       “key=value” format. The possible tokens are described
       below.
       When passing an option that OpenSIPS is not aware of, it
       will be blindly sent to the rtpengine daemon to be
       processed.
          + via-branch=... - Include the “branch” value of one of
            the “Via” headers in the request to the RTP proxy.
            Possible values are: “1” - use the first “Via” header;
            “2” - use the second “Via” header; “auto” - use the
            first “Via” header if this is a request, or the second
            one if this is a reply; “extra” - don't take the value
            from a header, but instead use the value of the
            “extra_id_pv” variable. This can be used to create one
            media session per branch on the RTP proxy. When
            sending a subsequent “delete” command to the RTP
            proxy, you can then stop just the session for a
            specific branch when passing the flag '1' or '2' in
            the “rtpengine_delete”, 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 the RTP proxy gets an “offer”
            command for a new branch, and then a “delete” command
            for the previous branch, which would otherwise delete
            the full call, breaking the subsequent “answer” for
            the new branch. This flag is only supported by the
            Sipwise rtpengine RTP proxy at the moment!
          + via-branch-param=... - provide a custom value for the
            via-branch param.
          + call-id - provide a custom Call-ID for the session. If
            missing, the Call-Id of the request/reply is used.
          + from-tag - provide a custom from-tag for the session.
            If missing, the from-tag request is used.
          + to-tag - provide a custom to-tag of the session. If
            missing, the to-tag of the request/reply is used, is
            present.
          + asymmetric - flags that UA from which message is
            received doesn't support symmetric RTP. (automatically
            sets the 'r' flag)
          + force-answer - force “answer”, 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.
          + in-iface=..., out-iface=... - these flags specify the
            direction the SIP message. These flags only make sense
            when the RTP proxy is running in bridge mode.
            “in-iface” should indicate the proxy's inbound
            interface, and “out-iface” corresponds to the RTP
            proxy's outbound interface. You always have to specify
            two flags to define the incoming network and the
            outgoing network. For example, “in-iface=internal
            out-iface=external” should be used for SIP message
            received from the local interface and sent out on the
            external interface.
          + internal, external - these the old flags used to
            specify the direction of call. They are now obsolate,
            being replaced by the “in-iface=internal
            out-iface=external” configuration.
          + auto-bridge - this flag an alternative to the
            “internal” and “external” flags in order to do
            automatic bridging between IPv4 on the "internal
            network" and IPv6 on the "external network". Instead
            of explicitly instructing the RTP proxy to select a
            particular address family, the distinction is done by
            the given IP in the SDP body by the RTP proxy itself.
            Not supported by Sipwise rtpengine.
          + address-family=... - instructs the RTP proxy that the
            recipient of this SDP body expects to see addresses of
            a particular family. Possible values are “IP4” and
            “IP6”. For example, if the SDP body contains IPv4
            addresses but the recipient only speaks IPv6, you
            would use “address-family=IP6” to bridge between the
            two address families.
            Sipwise rtpengine remembers the address family
            preference of each party after it has seen an SDP body
            from them. This means that normally it is only
            necessary to explicitly specify the address family in
            the “offer”, but not in the “answer”.
            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.
          + received-from=... - sets the address from which SIP
            packet with SDP received. This flag always set
            automatically, don't use it until you have a reason
            for that.
          + force - instructs the RTP proxy to ignore marks
            inserted by another RTP proxy in transit to indicate
            that the session is already goes through another
            proxy. Allows creating a chain of proxies. Not
            supported and ignored by Sipwise rtpengine.
          + trust-address - flags that IP address in SDP should be
            trusted. Without this flag, the RTP proxy ignores
            address in the SDP and uses source address of the SIP
            message as media address which is passed to the RTP
            proxy. From rtpengine 3.8 this is the default
            behaviour.
          + SIP-source-address - the opposite of trust-address.
            Restores the old default behaviour of ignoring
            endppoint of the addresses in the SDP body.
          + replace-origin - flags that IP from the origin
            description (o=) should be also changed.
          + replace-session-connection - flags to change the
            session-level SDP connection (c=) IP if media
            description also includes connection information.
          + replace-zero-address - flags to replace zero address
            with real address. Using a zero endpoint address is an
            obsolete way to signal a muted or sendonly stream.
            Streams with zero addresses are normally flagged as
            sendonly and the zero address in the SDP is passed
            through.
          + symmetric - flags that for the UA from which message
            is received, support symmetric RTP must be forced. You
            do not need to explicitly specify this value, as it is
            the default, and the behavior is only changed when the
            asymmetric is used.
          + repacketize=NN - requests the RTP proxy 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 RTP proxy 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. Not
            supported by Sipwise rtpengine.
          + loop-protect - flag that instructs RTP to avoid
            rewriting the SDP when looping the same message.
          + ICE=... - controls the RTP proxy's behaviour regarding
            ICE attributes within the SDP body. Possible values
            are: “force” - discard any ICE attributes already
            present in the SDP body and then generate and insert
            new ICE data, leaving itself as the only ICE
            candidates; “remove” instructs the RTP proxy to
            discard any ICE attributes and not insert any new ones
            into the SDP. The default (if no “ICE=...” is given at
            all), new ICE data will only be generated if no ICE
            was present in the SDP originally; otherwise the RTP
            proxy will only insert itself as an additional ICE
            candidate. Other SDP substitutions (c=, m=, etc) are
            unaffected by this flag.
          + RTP, SRTP, AVP, AVPF - These flags control the RTP
            transport protocol that should be used towards the
            recipient of the SDP. If none of them are specified,
            the protocol given in the SDP is left untouched.
            Otherwise, the “SRTP” flag indicates that SRTP should
            be used, while “RTP” indicates that SRTP should not be
            used. “AVPF” indicates that the advanced RTCP profile
            with feedback messages should be used, and “AVP”
            indicates that the regular RTCP profile should be
            used. See also the next set of flags below.
          + RTP/AVP, RTP/SAVP, RTP/AVPF, RTP/SAVPF - these serve
            as an alternative, more explicit way to select between
            the different RTP protocols and profiles supported by
            the RTP proxy. For example, giving the flag
            “RTP/SAVPF” has the same effect as giving the two
            flags “SRTP AVPF”.
          + to-tag - force inclusion of the “To” tag. Normally,
            the “To” tag is always included when present, except
            for “delete” messages. Including the “To” tag in a
            “delete” messages allows you to be more selective
            about which dialogues within a call are being torn
            down.
          + to-tag=... - use the specified string as “To” tag
            instead of the actual “To” tag from the SIP message,
            and force inclusion of the tag in the message as per
            above.
          + from-tag=... - use the specified string as “From” tag
            instead of the actual “From” tag from the SIP message.
          + call-id=... - use the specified string as “Call-ID”
            instead of the actual “Call-ID” from the SIP message.
          + rtcp-mux-demux - if rtcp-mux (RFC 5761) was offered,
            make the RTP proxy accept the offer, but not offer it
            to the recipient of this message.
          + rtcp-mux-reject - if rtcp-mux was offered, make the
            RTP proxy reject the offer, but still offer it to the
            recipient. Can be combined with “rtcp-mux-offer” to
            always offer it.
          + rtcp-mux-offer - make the RTP proxy offer rtcp-mux to
            the recipient of this message, regardless of whether
            it was offered originally or not.
          + rtcp-mux-require - Similar to offer but pretends that
            the client has accepted rtcp-mux. This breaks RFC 5761
            and will not advertise seperate RTCP ports. This
            option is necessary for WebRTC clients.
          + rtcp-mux-accept - if rtcp-mux was offered, make the
            RTP proxy accept the offer and also offer it to the
            recipient of this message. Can be combined with
            “rtcp-mux-offer” to always offer it.
          + media-address=... - force a particular media address
            to be used in the SDP body. Address family is detected
            automatically.
          + record-call=yes/no - indicates whether rtpengine
            should record the call or not. When using this
            parameter, you may pass further information in the
            “metadata”.
          + transcode-CODEC - used only for offer, indicates that
            rtpengine should transcode the CODEC towards the
            B-side. Example: transcode-PCMA will present to the
            B-side the PCMA codec.
          + codec-strip-CODEC - used only for offer, indicates
            that the A-side of the call will not end up talking
            CODEC. Example: codec-strip-PCMA will prevent the
            A-side from receiving the PCMA codec.
          + codec-mask-CODEC - used only for offer, indicates that
            the A-side will use the CODEC, but it will not be
            presented to the B-side. Example: codec-mask-PCMA will
            make the A-side receive the PCMA codec, but B-side
            will use something else.
     * sock_var(var, optional) - variable used to store the
       rtpengine socket chosen for this call.
     * sdp_var(var, optional) - variable used to store the full
       SDP received from rtpengine. You can perform any additional
       changes on this string. Important: when providing this
       variable, the message body is no longer changed, so you
       have to manually replace it!.
     * body(string, optional) - used to provide a specific body to
       the rtpengine_* function. If this parameter is missing the
       body of the current message is used.

   This function can be used from ALL_ROUTES.

   Example 1.15. rtpengine_offer usage
route {
...
    if (is_method("INVITE")) {
        if (has_body("application/sdp")) {
            if (rtpengine_offer())
                t_on_reply("1");
        } else {
            t_on_reply("2");
        }
    }
    if (is_method("ACK") && has_body("application/sdp"))
        rtpengine_answer();
...
}

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

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

   Example 1.16. rtpengine_offer usage with body replace
...
if (rtpengine_offer(, $var(socket), $var(body), $rb)) {
    xlog("Used rtpengine $var(socket)\n");
    # make all the changes on the resulted SDP in $var(body)
    ...
    remove_body_part();
    add_body_part($var(body), "application/sdp");
}
...

   Example 1.17. rtpengine_offer usage with call recording
...
$var(rtpengine_flags) = $var(rtpengine_flags) + " record-call=yes";

$json(recording_keys) := "{}";
$json(recording_keys/callId) = $ci;
$json(recording_keys/fromUser) = $dlg_val(recording_from_user);
$json(recording_keys/fromDomain) = $dlg_val(recording_from_domain);
$json(recording_keys/fromTag) = $dlg_val(recording_from_tag);
$json(recording_keys/toUser) = $dlg_val(recording_to_user);
$json(recording_keys/toDomain) = $dlg_val(recording_to_domain);

$var(rtpengine_flags) = $var(rtpengine_flags) + " metadata=" + $(json(re
cording_keys){s.encode.hexa});
rtpengine_offer($var(rtpengine_flags));
...

   Example 1.18. rtpengine_offer usage for transcoding
...
# Goal: make A-side talk PCMA and B-side talk opus
# * do not present PCMA to B-side: codec-mask-PCMA, but use it on A-side
# * do not use opus for A-side: codec-strip-opus
# * offer opus to B-side: transcode-opus
rtpengine_offer("... codec-mask-PCMA codec-strip-opus transcode-opus ...
");
...

1.5.3.  rtpengine_answer([flags[, sock_pvar[, sdp_pvar[, body]]]])

   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 rtpengine_offer() function description above for the
   meaning of the parameters.

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

   Example 1.19. rtpengine_answer usage

   See rtpengine_offer() function example above for examples.

1.5.4.  rtpengine_delete([flags[, sock_var]])

   Tears down the RTPEngine session for the current call.

   See rtpengine_offer() function description above for the
   meaning of the parameters. Note that not all flags make sense
   for a “delete”.

   This function can be used from ALL_ROUTES.

   Example 1.20. rtpengine_delete usage
...
rtpengine_delete();
...

1.5.5.  rtpengine_manage([flags[, sock_var[, sdp_var[, body]]]])

   Manage the RTPEngine session - it combines the functionality of
   rtpengine_offer(), rtpengine_answer() and rtpengine_delete(),
   detecting internally based on message type and method which one
   to execute.

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

   Functionality:
     * If INVITE with SDP, then do rtpengine_offer()
     * If ACK with SDP, then do rtpengine_answer()
     * If BYE or CANCEL, or called within a FAILURE_ROUTE[], then
       do rtpengine_delete()
     * If reply to INVITE with code >= 300 do rtpengine_delete()
     * If reply with SDP to INVITE having code 1xx and 2xx, then
       do rtpengine_answer() if the request had SDP or tm is not
       loaded, otherwise do rtpengine_offer()

   This function can be used from ALL_ROUTES.

   Example 1.21. rtpengine_manage usage
...
rtpengine_manage();
...

1.5.6.  rtpengine_start_recording([flags [, sock_var]])

   This function will send a signal to the RTP proxy to record the
   RTP stream on the RTP proxy.

   Meaning of the parameters is as follows:
     * flags(string, optional) - flags used to change the behavior
       of the recorder. An importat value to set is the call-id
       value, which can be used to start recording a different
       call than the requested one.
     * sock_var(var, optional) - variable used to store the
       rtpengine socket chosen for this call.

   This function can be used from any route.

   Example 1.22. rtpengine_start_recording usage
...
rtpengine_start_recording();
...

1.5.7.  rtpengine_stop_recording([flags [, sock_var]])

   This function will send a signal to the RTP proxy to stop
   recording the RTP stream on the RTP proxy.

   Meaning of the parameters is as follows:
     * flags(string, optional) - flags used to change the behavior
       of the recorder. An importat value to set is the call-id
       value, which can be used to start recording a different
       call than the requested one.
     * sock_var(var, optional) - variable used to store the
       rtpengine socket chosen for this call.

   This function can be used from any route.

   Example 1.23. rtpengine_stop_recording usage
...
rtpengine_stop_recording();
...

1.5.8.  rtpengine_play_media(flags, [duration_spec[, sock_var[,
sockvar]]])

   This function will start playing a media file to one of the
   endpoints.

   Meaning of the parameters is as follows:
     * flags(string) - a list of flags simialar to the other
       functions. One of the file, blob or db-id parameters is
       mandatory to indicate the content of the media file to be
       played. file is a common choice for specifying rtpengine to
       get media from a file path, blob to take the content from
       an inline string and db-id to get the content from the
       database.
       The direction of the media stream is controlled by the
       from-tag parameter, address (media address from the SDP),
       or label, if the media stream contains a label. If all of
       them are missing, the media file is played to the initiator
       of the SIP request, and will work similar to a ringback
       tone.
     * duration_spec(var, optional) - a pseudo variable that will
       contain the duration of the played file. It will be set to
       -1 if the duration could not be determined.
     * sock_var(var, optional) - variable used to store the
       rtpengine socket chosen for this call.

   This function can be used from any route.

   Example 1.24. Ringback tone using rtpengine_play_media
...
if (is_method("INVITE") && !has_totag())
        rtpengine_play_media("file=/path/to/ringback_tone_file.wav");
...

   Example 1.25. Manage music on hold using rtpengine_play_media
...
if (is_method("INVITE") && has_totag()) {
        if (is_audio_on_hold()) {
                $dlg_val(on_hold) = "1";
                rtpengine_play_media("from-tag=$tt file=/path/to/moh_fil
e.wav");
        } else if ($dlg_val(on_hold) == "1") {
                $dlg_val(on_hold) = "0";
                rtpengine_stop_media("from-tag=$tt");
        }
}
...

1.5.9.  rtpengine_stop_media([flags[, sockvar]])

   This function will stop playing a media file previously started
   by a rtpengine_play_media() call. The meaning of its parameters
   is similar to the previous functions. Note that this function
   should be called with similar parameters as its matching
   rtpengine_play_media() call, otherwise RTPEngine will not be
   able to stop media playing.

   This function can be used from any route.

   Example 1.26. Ringback tone stop using rtpengine_stop_media
...
if (is_method("INVITE") && $rs == 200)
        rtpengine_stop_media();
...

1.5.10.  rtpengine_block_media([flags[, sockvar]])

   This function will block the media sent from one of the
   endpoints. The direction to be blocked is controled by the
   flags parameter, the from-tag value.

   This function can be used from any route.

   Example 1.27. Example of rtpengine_block_media usage
...
rtpengine_block_media();
...

1.5.11.  rtpengine_unblock_media([flags[, sockvar]])

   This function will resume/unblock the media sent from one of
   the endpoints. The direction to be blocked is controled by the
   flags parameter, the from-tag value.

   This function can be used from any route.

   Example 1.28. Example of rtpengine_unblock_media usage
...
rtpengine_unblock_media();
...

1.5.12.  rtpengine_block_dtmf([flags[, sockvar]])

   This function will block the DTMF media sent from one of the
   endpoints. The direction to be blocked is controled by the
   flags parameter, the from-tag value.

   This function can be used from any route.

   Example 1.29. Example of rtpengine_block_dtmf usage
...
rtpengine_block_dtmf();
...

1.5.13.  rtpengine_unblock_dtmf([flags[, sockvar]])

   This function will resume/unblock the DTMF media sent from one
   of the endpoints. The direction to be blocked is controled by
   the flags parameter, the from-tag value.

   This function can be used from any route.

   Example 1.30. Example of rtpengine_unblock_dtmf usage
...
rtpengine_unblock_dtmf();
...

1.5.14.  rtpengine_start_forwarding([flags[, sockvar]])

   This function will start forwarding the media to a TLS
   destination specified in the tls-send-to parmeter of RTPEngine.
   This function allows you to select the media stream to forward,
   by specifing the from-tag of the entity you want to forward the
   media. If missing, all media streams are forwarded.

   This function can be used from any route.

   Example 1.31. Example of rtpengine_start_forwarding usage
...
rtpengine_start_forwarding();
...

1.5.15.  rtpengine_stop_forwarding([flags[, sockvar]])

   This function will stop forwarding of the media previously
   started using the rtpengine_start_forwarding() function.

   This function can be used from any route.

   Example 1.32. Example of rtpengine_stop_forwarding usage
...
rtpengine_stop_forwarding();
...

1.5.16.  rtpengine_play_dtmf(code, [flags[, sockvar]])

   This function instructs RTP to send the DTMF code to the
   participant of the call. The code can be a digit (“0-9”) or a
   special character (one of “*,#,A,B,C,D”). Additional parameters
   can be configured using the flags parameter. For more
   information, please consult the RTP documentation.

   NOTE: if you are planning to inject DTMF in a session, you have
   to specify the inject-DTMF flag when the session is created.

   This function can be used to convert SIP INFO DTMF keys to RTP
   DTMF.

   This function can be used from any route.

   Example 1.33. Example of rtpengine_play_dtmf usage
...
rtpengine_play_dtmf("0"); # send the 0 code upstream
...

1.6. Exported Pseudo-Variables

1.6.1. $rtpstat

   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.

   Example 1.34. $rtpstat Usage
...
    append_hf("X-RTP-Statistics: $rtpstat\r\n");
...

1.6.2. $rtpstat(STAT)[index]

   Returnes one of the pre-fined statistics listed below:
     * MOS-average - without an index, it returns the average MOS
       value, expressed in an integer between 0 and 50, of all the
       RTP streams involved in the call, both caller and callee.
       If index is specified, it has to be one of the from-tag or
       to-tag involved in the call. In this case, the variable
       will return the average MOS of all the streams generated by
       that endpoint with the associated tag value. If you need
       more granular statistics, check the $rtpquery variable.
     * jitter-average - similar behavior with MOS-average, but
       returnes the average jitter.
     * roundtrip-average - similar behavior with MOS-average, but
       returnes the average roundtrip.
     * packetloss-average - similar behavior with MOS-average, but
       returnes the average packet loss.
     * MOS-min - without an index, it returns the minimum MOS
       value (integer value between 0 and 50) of all RTP streams
       involved in the call, both caller and callee. If the index
       is specified, it has the same effect as for MOS-average.
     * jitter-min - similar behavior with MOS-min, but returnes
       the minimum jitter of a leg/call.
     * roundtrip-min - similar behavior with MOS-min, but returnes
       the minimum roundtrip of a leg/call.
     * packetloss-min - similar behavior with MOS-min, but
       returnes the minimum packet loss of a leg/call.
     * MOS-max - without an index, it returns the maximum MOS
       value (integer value between 0 and 50) of all RTP streams
       involved in the call, both caller and callee. If the index
       is specified, it has the same effect as for MOS-average.
     * jitter-max - similar behavior with MOS-max, but returnes
       the maximum jitter of a leg/call.
     * roundtrip-max - similar behavior with MOS-max, but returnes
       the maximum roundtrip of a leg/call.
     * packetloss-max - similar behavior with MOS-max, but
       returnes the maximum packet loss of a leg/call.
     * MOS-min-at - without an index, it returns the time in
       seconds elapsed from the start of the call when the MOS
       value is minimum. If the index is specified, it has the
       same effect as for MOS-average.
     * jitter-min-at - similar behavior with MOS-min-at, but
       returnes the time when the minimum jitter was detected.
     * roundtrip-min-at - similar behavior with MOS-min-at, but
       returnes the time when the minimum roundtrip was detected.
     * packetloss-min-at - similar behavior with MOS-min-at, but
       returnes the time when the minimum packet loss of a
       leg/call was detected.
     * MOS-max-at - without an index, it returns the time in
       seconds elapsed from the start of the call when the MOS
       value is maximum. If the index is specified, it has the
       same effect as for MOS-average.
     * jitter-max-at - similar behavior with MOS-max-at, but
       returnes the time when the maximum value of jitter was
       detected.
     * roundtrip-max-at - similar behavior with MOS-max-at, but
       returnes the time when the maximum value of roundtrip was
       detected.
     * packetloss-min-at - similar behavior with MOS-max-at, but
       returnes the time when the maximum packet loss of a
       leg/call was detected.

   NOTE: all these statistics are computed based on the statistics
   generated by RTPEngine. Some of them might not be available for
   all the calls (i.e. MOS cannot be computed if the call is too
   short, or if the phones do not properly report RTP statistics
   over RTCP). In these cases the variable returns the NULL value.

   Example 1.35. $rtpstat(STAT)
...
    xlog("Average MOS of the entire call is $rtpstat(MOS-average)\r\n");
    xlog("Average MOS of caller is $(rtpstat(MOS-average)[$ft])\r\n");
    xlog("Average MOS of callee is $(rtpstat(MOS-average)[$tt])\r\n");
    xlog("Min MOS of caller is $(rtpstat(MOS-min)[$ft]) reported at $(rt
pstat(MOS-min-at)[$ft])\r\n");
...

1.6.3. $rtpquery

   Does a Query command to the RTP proxy and returns the answer in
   a JSON format. You can use this variable to fetch arbitrary
   data from the RTP proxy such as raw statistics about the call,
   or other indicators.

   You can use a $json() variable to parse its output and extract
   any information from the query, such as RTP statistics, or MOS
   values.

   Example 1.36. $rtpquery Usage
...
        $json(reply) := $rtpquery;
        xlog("Total RTP Stats: $json(reply/totals)\n");
...

1.7. Exported MI Functions

1.7.1. rtpengine_enable

   Enables/disables a RTP proxy.

   Parameters:
     * url - the RTP proxy url (exactly as defined in the config
       file).
     * enable - 1 - enable, 0 - disable the RTP proxy.
     * setid (optional) the set ID of the nodes to be updated. If
       provided, only nodes in the provided set will be updated.

   NOTE: if a RTP proxy is defined multiple times (in the same or
   different set), all of its instances will be enabled/disabled
   IF no set ID is provided.

   Example 1.37.  rtpengine_enable usage
...
## disable all rtpengines by URL
$ opensips-cli -x mi rtpengine_enable udp:192.168.2.133:8081 0
## enable rtpengine by URL and set ID (3)
$ opensips-cli -x mi rtpengine_enable url=udp:192.168.2.133:8081 enable=
1 setid=3
...

1.7.2. rtpengine_show

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

   No parameter.

   Example 1.38.  rtpengine_show usage
...
$ opensips-cli -x mi rtpengine_show
...

1.7.3. rtpengine_reload

   Reloads all rtpengine sets from the database. Used only when
   the “db_url” parameter is set.

   Parameters:
     * type (optional) soft - when reloading nodes from the
       database, reuse any existing sockets and keep existing node
       disabled state. If not provided, then all nodes and sockets
       will first be torndown and then nodes will be loaded from
       the database.

   No parameter.

   Example 1.39.  rtpengine_reload usage
...
$ opensips-cli -x mi rtpengine_reload
$ opensips-cli -x mi rtpengine_reload type=soft
...

1.7.4. teardown

   Terminates the SIP dialog by the SIP Call-ID given as
   parameter.

   Parameters:
     * callid - SIP Call-ID.

   Note this is a just a wrapper function over the “dlg_end_dlg”
   MI function provided by the “dialog” module. This wrapping is
   done just to make rtpengine happy when trying to terminate SIP
   calls based on RTP timeouts.

   Example 1.40.  teardown usage
...
$ opensips-cli -x mi teardown Y2IwYjQ2YmE2ZDg5MWVkNDNkZGIwZjAzNGM1ZDY0ZD
Q
...

1.8. Exported Events

1.8.1.  E_RTPENGINE_NOTIFICATION

   This event is raised when a notification is received from
   RTPengine.

   Parameters represent the nodes within the Json request received
   from RTPengine. Common values are:
     * type - identifies the type of notification (i.e. DTMF)
     * callid - the callid of the call this event is triggered for
     * source_tag - from tag of the call this event is triggered
       for
     * timestamp - timestamp when the event was triggered

   For a DTMF event received, you will also get the following
   nodes:
     * source_ip - the IP that triggered the DTMF
     * event - the event/digit pressed
     * duration - how long the digit was pressed
     * volume - volume of the tone

1.8.2.  E_RTPENGINE_STATUS

   This event is raised when a RTPEngine server changes it's
   status to active/inactive.

   Parameters:
     * socket - the socket that identifies the RTPEngine instance.
     * status - active if the RTPEngine instance responds to
       probing or inactive if the instance was deactivated.

Chapter 2. Frequently Asked Questions

   2.1.

   How do I migrate from “rtpproxy” or “rtpproxy-ng” to
   “rtpengine”?

   For the most part, only the names of the functions have
   changed, with “rtpproxy” in each name replaced with
   “rtpengine”. For example, “rtpproxy_manage()” has become
   “rtpengine_manage()”. A few name duplications have also been
   resolved, for example there is now a single
   “rtpengine_delete()” instead of “unforce_rtp_proxy()” and the
   identical “rtpproxy_destroy()”.

   The largest difference to the old module is how flags are
   passed to “rtpengine_offer()”, “rtpengine_answer()”,
   “rtpengine_manage()” and “rtpengine_delete()”. Instead of
   having a string of single-letter flags, they now take a string
   of space-separated items, with each item being either a single
   token (word) or a “key=value” pair.

   For example, if you had a call “rtpproxy_offer("FRWOC+PS");”,
   this would then become:
rtpengine_offer("force trust-address symmetric replace-origin replace-se
ssion-connection ICE=force RTP/SAVPF");

   Finally, if you were using the second parameter (explicit media
   address) to any of these functions, this has been replaced by
   the “media-address=...” option within the first string of
   flags.

   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) 231 129 6059 3095
   2. Bogdan-Andrei Iancu (@bogdan-iancu) 31 17 423 595
   3. John Burke (@john08burke) 25 17 647 102
   4. Liviu Chircu (@liviuchircu) 20 16 91 173
   5. Richard Fuchs 20 2 640 723
   6. Vlad Patrascu (@rvlad-patrascu) 15 7 218 330
   7. Peter Lemenkov (@lemenkov) 10 7 27 62
   8. Eric Tamme (@etamme) 7 5 42 19
   9. Nick Altmann (@nikbyte) 6 4 43 2
   10. Maksym Sobolyev (@sobomax) 6 4 11 12

   All remaining contributors: Ovidiu Sas (@ovidiusas), Zero King
   (@l2dy), Rob Gagnon (@rgagnon24), Flavio E. Goncalves, Dan
   Pascu (@danpascu), Oliver Severin Mulelid-Tynes (@olivermt).

   (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.  Razvan Crainea (@razvancrainea)     Jun 2014 - Aug 2025
   2.  Maksym Sobolyev (@sobomax)          Jan 2021 - Nov 2023
   3.  Liviu Chircu (@liviuchircu)         Jul 2014 - May 2023
   4.  Peter Lemenkov (@lemenkov)          Jun 2018 - Apr 2022
   5.  John Burke (@john08burke)           Jun 2019 - Apr 2022
   6.  Bogdan-Andrei Iancu (@bogdan-iancu) Jun 2014 - Apr 2022
   7.  Nick Altmann (@nikbyte)             May 2021 - May 2021
   8.  Flavio E. Goncalves                 Oct 2020 - Oct 2020
   9.  Zero King (@l2dy)                   Mar 2020 - Sep 2020
   10. Ovidiu Sas (@ovidiusas)             Jun 2020 - Jun 2020

   All remaining contributors: Vlad Patrascu (@rvlad-patrascu),
   Dan Pascu (@danpascu), Oliver Severin Mulelid-Tynes
   (@olivermt), Rob Gagnon (@rgagnon24), Eric Tamme (@etamme),
   Richard Fuchs.

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

Chapter 4. Documentation

4.1. Contributors

   Last edited by: Razvan Crainea (@razvancrainea), Liviu Chircu
   (@liviuchircu), John Burke (@john08burke), Nick Altmann
   (@nikbyte), Flavio E. Goncalves, Peter Lemenkov (@lemenkov),
   Vlad Patrascu (@rvlad-patrascu), Bogdan-Andrei Iancu
   (@bogdan-iancu), Richard Fuchs.

   Documentation Copyrights:

   Copyright © 2013-2014 Sipwise GmbH

   Copyright © 2010 VoIPEmbedded Inc.

   Copyright © 2009-2014 TuTPro Inc.

   Copyright © 2005 Voice Sistem SRL

   Copyright © 2003-2008 Sippy Software, Inc.
