rtpproxy Module

Maxim Sobolev

   Sippy Software, Inc.

Edited by

Maxim Sobolev

Edited by

Bogdan-Andrei Iancu

   Copyright © 2003-2008 Sippy Software, Inc.

   Copyright © 2005 Voice Sistem SRL
   Revision History
   Revision $Revision: 8740 $ $Date$
     __________________________________________________________

   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. nortpproxy_str (string)
              1.5.8. db_url (string)
              1.5.9. db_table (string)
              1.5.10. rtpp_socket_col (string)
              1.5.11. set_id_col (string)
              1.5.12. rtpp_notify_socket (string)
              1.5.13. rtpp_sock_pvar (string)

        1.6. Exported Functions

              1.6.1. set_rtp_proxy_set()
              1.6.2. engage_rtp_proxy([flags [, ip_address]])
              1.6.3. rtpproxy_offer([flags [, ip_address]])
              1.6.4. rtpproxy_answer([flags [, ip_address]])
              1.6.5. unforce_rtp_proxy()
              1.6.6. rtpproxy_stream2uac(prompt_name, count),
                      rtpproxy_stream2uas(prompt_name, count)

              1.6.7. rtpproxy_stop_stream2uac(),
                      rtpproxy_stop_stream2uas()

              1.6.8. start_recording()

        1.7. MI Commands

              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

   2. Frequently Asked Questions

   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 nortpproxy_str parameter
   1.7. Set db_url parameter
   1.8. Set db_table parameter
   1.9. Set rtpp_socket_col parameter
   1.10. Set set_id parameter
   1.11. Set rtpp_notify_socket parameter
   1.12. Set rtpp_sock_pvar parameter
   1.13. fix_nated_contact usage
   1.14. engage_rtp_proxy usage
   1.15. rtpproxy_offer usage
   1.16. rtpproxy_answer usage
   1.17. unforce_rtp_proxy usage
   1.18. rtpproxy_stream2xxx usage
   1.19. start_recording usage
   1.20. rtpproxy_enable usage
   1.21. rtpproxy_show usage
   1.22. 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.

   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()!!

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.

   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
   engage_rtp_proxy(), 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:
   http://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 engage_rtp_proxy 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.

   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")
...

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

1.5.8. 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 1.7. Set db_url parameter
...
modparam("rtpproxy", "db_url",
                "mysql://opensips:opensipsrw@192.168.2.132/opensips")
...


1.5.9. 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 1.8. Set db_table parameter
...
modparam("rtpproxy", "db_table", "nh_sockets")
...


1.5.10. rtpp_socket_col (string)

   The name rtpp socket column in the database table.

   Default value is “rtpproxy_sock”.

   Example 1.9. Set rtpp_socket_col parameter
...
modparam("rtpproxy", "rtpp_socket_col", "rtpp_socket")
...


1.5.11. set_id_col (string)

   The name set id column in the database table.

   Default value is “set_id”.

   Example 1.10. Set set_id parameter
...
modparam("rtpproxy", "set_id_col", "rtpp_set_id")
...


1.5.12. rtpp_notify_socket (string)

   The socket used by OpenSIPS to receive timeout notifications.

   Default value is “NULL”.

   Example 1.11. Set rtpp_notify_socket parameter
...
modparam("rtpproxy", "rtpp_notify_socket", "tcp:10.10.10.10:9999")
...


1.5.13. rtpp_sock_pvar (string)

   Socket url for the rtpproxy instance used for setting up the
   call.

   Default value is “NULL (don't set pvar)”.

   Example 1.12. Set rtpp_sock_pvar parameter
...
modparam("rtpproxy", "rtpp_sock_pvar", "$avp(rtp_sock)")

#Load balancing rtpproxies
modparam("rtpproxy", "rtpproxy_sock",
        "udp:localhost:12221 udp:localhost:12222")

engage_rtp_proxy("roc");
xlog("Used rtpproxy $avp(rtp_sock) for callid $ci");
...


1.6. Exported Functions

1.6.1.  set_rtp_proxy_set()

   Sets the Id of the rtpproxy set to be used for the next
   unforce_rtp_proxy(), rtpproxy_offer() or rtpproxy_answer()
   command.

   Paramter can also be a pseudo-variable that contain (as string
   or integer) the ID of the rtpproxy set to be used.

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

   Example 1.13. fix_nated_contact usage
...
set_rtp_proxy_set("2");
rtpproxy_offer();
...

1.6.2.  engage_rtp_proxy([flags [, ip_address]])

   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 - flags to turn on some features.
          + a - flags that UA from which message is received
            doesn't support symmetric RTP.
          + l - force “lookup”, that is, only rewrite SDP when
            corresponding session is already exists in the RTP
            proxy. By default is on when the session is to be
            completed (reply in non-swap or ACK in swap mode).
          + i/e - when RTPProxy is used in bridge mode, these
            flags are used to indicate the direction of the media
            flow for the current request/reply. 'i' refers to the
            LAN (internal network) and corresponds to the first
            interface of RTPProxy (as specified by the -l
            parameter). 'e' refers to the WAN (external network)
            and corresponds to the second interface of RTPProxy.
            These flags should always be used together. For
            example, an INVITE (offer) that comes from the
            Internet (WAN) to goes to a local media server (LAN)
            should use the 'ei' flags. The answer should use the
            'ie' flags. Depending on the scenario, the 'ii' and
            'ee' combination are also supported. Only makes sense
            when RTPProxy is running in the bridge mode.
          + f - instructs rtpproxy to ignore marks inserted by
            another rtpproxy in transit to indicate that the
            session is already goes through another proxy. Allows
            creating chain of proxies.
          + r - flags that IP address in SDP should be trusted.
            Without this flag, rtpproxy ignores address in the SDP
            and uses source address of the SIP message as media
            address which is passed to the RTP proxy.
          + o - flags that IP from the origin description (o=)
            should be also changed.
          + c - flags to change the session-level SDP connection
            (c=) IP if media-description also includes connection
            information.
          + s/w - flags that for the UA from which message is
            received, support symmetric RTP must be forced.
          + n - flags that enables the notification timeout for
            the session.
          + zNN - requests the RTPproxy to perform
            re-packetization of RTP traffic coming from the UA
            which has sent the current message to increase or
            decrease payload size per each RTP packet forwarded if
            possible. The NN is the target payload size in ms, for
            the most codecs its value should be in 10ms
            increments, however for some codecs the increment
            could differ (e.g. 30ms for GSM or 20ms for G.723).
            The RTPproxy would select the closest value supported
            by the codec. This feature could be used for
            significantly reducing bandwith overhead for low
            bitrate codecs, for example with G.729 going from 10ms
            to 100ms saves two thirds of the network bandwith.
     * ip_address - new SDP IP address.

   Both parmeters do accept variables in any combination.

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

   Example 1.14. engage_rtp_proxy usage
                ...
                if (is_method("INVITE") &&
                has_totag()) {engage_rtp_proxy();};
                ...

1.6.3.  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.

   See engage_rtp_proxy() 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_offer usage
route {
...
    if (is_method("INVITE")) {
        if (has_sdp()) {
            if (rtpproxy_offer())
                t_on_reply("1");
        } else {
            t_on_reply("2");
        }
    }
    if (is_method("ACK") && has_sdp())
        rtpproxy_answer();
...
}

onreply_route[1]
{
...
    if (has_sdp())
        rtpproxy_answer();
...
}

onreply_route[2]
{
...
    if (has_sdp())
        rtpproxy_offer();
...
}

1.6.4.  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 engage_rtp_proxy() function description above for the
   meaning of the parameters.

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

   Example 1.16. rtpproxy_answer usage

   See rtpproxy_offer() function example above for example.

1.6.5.  unforce_rtp_proxy()

   Tears down the RTPProxy session for the current call.

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

   Example 1.17. unforce_rtp_proxy usage
...
unforce_rtp_proxy();
...

1.6.6.  rtpproxy_stream2uac(prompt_name, count),
rtpproxy_stream2uas(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, functions require that the session
   in the RTPproxy already exists. Also those functions don't
   alted SDP, so that they are not substitute for calling
   rtpproxy_offer or rtpproxy_answer.

   This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE.

   Meaning of the parameters is as follows:
     * prompt_name - name of the prompt to stream. Should be
       either absolute pathname or pathname relative to the
       directory where RTPproxy runs.
     * count - number of times the prompt should be repeated. The
       value of -1 means that it will be streaming in loop
       indefinitely, until appropriate rtpproxy_stop_stream2xxx is
       issued.

   Example 1.18. 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();
        };
    };
...

1.6.7.  rtpproxy_stop_stream2uac(), rtpproxy_stop_stream2uas()

   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.

1.6.8.  start_recording()

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

   This function can be used from REQUEST_ROUTE and ONREPLY_ROUTE.

   Example 1.19. start_recording usage
...
start_recording();
...

1.7. MI Commands

1.7.1. 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
   diferente sete), all its instances will be enables/disabled.

   Example 1.20.  rtpproxy_enable usage
...
$ opensipsctl fifo rtpproxy_enable udp:192.168.2.133:8081 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 1.21.  rtpproxy_show usage
...
$ opensipsctl fifo 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 1.22.  rtpproxy_reload usage
...
$ opensipsctl fifo 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.

Chapter 2. Frequently Asked Questions

   2.1.

       What happend 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 http://www.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.
