rtpproxy Module

Maxim Sobolev

   Sippy Software, Inc.

Juha Heinanen

   TuTPro, Inc.

Edited by

Maxim Sobolev

Edited by

Bogdan-Andrei Iancu

Edited by

Juha Heinanen

Edited by

Sas Ovidiu

Edited by

Carsten Bock

   ng-voice GmbH

   Copyright © 2003-2008 Sippy Software, Inc.

   Copyright © 2005 Voice Sistem SRL

   Copyright © 2009-2012 TuTPro Inc.

   Copyright © 2010 VoIPEmbedded Inc.
     __________________________________________________________________

   Table of Contents

   1. Admin Guide

        1. Overview
        2. Multiple RTPProxy usage
        3. Dependencies

              3.1. Kamailio Modules
              3.2. External Libraries or Applications

        4. Parameters

              4.1. rtpproxy_sock (string)
              4.2. rtpproxy_disable_tout (integer)
              4.3. rtpproxy_tout (integer)
              4.4. rtpproxy_retr (integer)
              4.5. nortpproxy_str (string)
              4.6. timeout_socket (string)
              4.7. ice_candidate_priority_avp (string)
              4.8. extra_id_pv (string)

        5. Functions

              5.1. set_rtp_proxy_set(setid)
              5.2. rtpproxy_offer([flags [, ip_address]])
              5.3. rtpproxy_answer([flags [, ip_address]])
              5.4. rtpproxy_destroy([flags])
              5.5. unforce_rtp_proxy()
              5.6. rtpproxy_manage([flags [, ip_address]])
              5.7. rtpproxy_stream2uac(prompt_name, count),
              5.8. rtpproxy_stream2uas(prompt_name, count)
              5.9. rtpproxy_stop_stream2uac(),
              5.10. start_recording()
              5.11. rtpproxy_stop_stream2uas(prompt_name, count)

        6. Exported Pseudo Variables

              6.1. $rtpstart

        7. MI Commands

              7.1. nh_enable_rtpp
              7.2. nh_show_rtpp

   2. Frequently Asked Questions

   List of Examples

   1.1. Set rtpproxy_sock parameter
   1.2. Set rtpproxy_disable_tout parameter
   1.3. Set rtpproxy_tout parameter
   1.4. Set rtpproxy_retr parameter
   1.5. Set nortpproxy_str parameter
   1.6. Set timeout_socket parameter
   1.7. Set ice_candidate_priority_avp parameter
   1.8. Set extra_id_pv parameter
   1.9. set_rtp_proxy_set usage
   1.10. rtpproxy_offer usage
   1.11. rtpproxy_answer usage
   1.12. rtpproxy_destroy usage
   1.13. rtpproxy_manage usage
   1.14. rtpproxy_stream2xxx usage
   1.15. start_recording usage
   1.16. $rtpstat-Usage
   1.17. nh_enable_rtpp usage
   1.18. nh_show_rtpp usage

Chapter 1. Admin Guide

   Table of Contents

   1. Overview
   2. Multiple RTPProxy usage
   3. Dependencies

        3.1. Kamailio Modules
        3.2. External Libraries or Applications

   4. Parameters

        4.1. rtpproxy_sock (string)
        4.2. rtpproxy_disable_tout (integer)
        4.3. rtpproxy_tout (integer)
        4.4. rtpproxy_retr (integer)
        4.5. nortpproxy_str (string)
        4.6. timeout_socket (string)
        4.7. ice_candidate_priority_avp (string)
        4.8. extra_id_pv (string)

   5. Functions

        5.1. set_rtp_proxy_set(setid)
        5.2. rtpproxy_offer([flags [, ip_address]])
        5.3. rtpproxy_answer([flags [, ip_address]])
        5.4. rtpproxy_destroy([flags])
        5.5. unforce_rtp_proxy()
        5.6. rtpproxy_manage([flags [, ip_address]])
        5.7. rtpproxy_stream2uac(prompt_name, count),
        5.8. rtpproxy_stream2uas(prompt_name, count)
        5.9. rtpproxy_stop_stream2uac(),
        5.10. start_recording()
        5.11. rtpproxy_stop_stream2uas(prompt_name, count)

   6. Exported Pseudo Variables

        6.1. $rtpstart

   7. MI Commands

        7.1. nh_enable_rtpp
        7.2. nh_show_rtpp

1. Overview

   This is a module that enables media streams to be proxied via an
   rtpproxy. Rtpproxies know to work with this module are Sippy RTPproxy
   http://www.rtpproxy.org and ngcp-rtpproxy-ng
   http://deb.sipwise.com/spce/2.6/pool/main/n/ngcp-mediaproxy-ng. Some
   features of the rtpproxy module apply only to one of the two
   rtpproxies.

2. Multiple RTPProxy usage

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

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

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

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

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

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

3. Dependencies

   3.1. Kamailio Modules
   3.2. External Libraries or Applications

3.1. Kamailio Modules

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

3.2. External Libraries or Applications

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

4. Parameters

   4.1. rtpproxy_sock (string)
   4.2. rtpproxy_disable_tout (integer)
   4.3. rtpproxy_tout (integer)
   4.4. rtpproxy_retr (integer)
   4.5. nortpproxy_str (string)
   4.6. timeout_socket (string)
   4.7. ice_candidate_priority_avp (string)
   4.8. extra_id_pv (string)

4.1. 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
modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:12221")
# multiple rtproxies for LB
modparam("rtpproxy", "rtpproxy_sock",
        "udp:localhost:12221 udp:localhost:12222")
# multiple sets of multiple rtproxies
modparam("rtpproxy", "rtpproxy_sock",
        "1 == udp:localhost:12221 udp:localhost:12222")
modparam("rtpproxy", "rtpproxy_sock",
        "2 == udp:localhost:12225")
...

4.2. rtpproxy_disable_tout (integer)

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

   Default value is “60”.

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

4.3. rtpproxy_tout (integer)

   Timeout value in waiting for reply from RTPProxy.

   Default value is “1”.

   Example 1.3. Set rtpproxy_tout parameter
...
modparam("rtpproxy", "rtpproxy_tout", 2)
...

4.4. rtpproxy_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 rtpproxy_retr parameter
...
modparam("rtpproxy", "rtpproxy_retr", 2)
...

4.5. nortpproxy_str (string)

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

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

4.6. timeout_socket (string)

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

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

   Default value is “” (nothing).

   Example 1.6. Set timeout_socket parameter
...
modparam("nathelper", "timeout_socket", "xmlrpc:http://127.0.0.1:8000/RPC2")
...

4.7. ice_candidate_priority_avp (string)

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

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

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

   Example 1.7. Set ice_candidate_priority_avp parameter
...
modparam("rtpproxy", "ice_candidate_priority_avp", "$avp(ice_priority)")
...

4.8. extra_id_pv (string)

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

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

   Example 1.8. Set extra_id_pv parameter
...
modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)")
...

5. Functions

   5.1. set_rtp_proxy_set(setid)
   5.2. rtpproxy_offer([flags [, ip_address]])
   5.3. rtpproxy_answer([flags [, ip_address]])
   5.4. rtpproxy_destroy([flags])
   5.5. unforce_rtp_proxy()
   5.6. rtpproxy_manage([flags [, ip_address]])
   5.7. rtpproxy_stream2uac(prompt_name, count),
   5.8. rtpproxy_stream2uas(prompt_name, count)
   5.9. rtpproxy_stop_stream2uac(),
   5.10. start_recording()
   5.11. rtpproxy_stop_stream2uas(prompt_name, count)

5.1.  set_rtp_proxy_set(setid)

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

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

   Example 1.9. set_rtp_proxy_set usage
...
set_rtp_proxy_set("2");
rtpproxy_offer();
...

5.2.  rtpproxy_offer([flags [, ip_address]])

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

   Meaning of the parameters is as follows:
     * flags - flags to turn on some features.
          + 1 - append first Via branch to Call-ID when sending command to
            rtpproxy. This can be used to create one media session per
            branch on the rtpproxy. When sending a subsequent “delete”
            command to the rtpproxy, you can then stop just the session
            for a specific branch when passing the flag '1' or '2' in the
            “unforce_rtpproxy”, or stop all sessions for a call when not
            passing one of those two flags there. This is especially
            useful if you have serially forked call scenarios where
            rtpproxy gets an “update” command for a new branch, and then a
            “delete” command for the previous branch, which would
            otherwise delete the full call, breaking the subsequent
            “lookup” for the new branch. This flag is only supported by
            the ngcp-mediaproxy-ng rtpproxy at the moment!
          + 2 - append second Via branch to Call-ID when sending command
            to rtpproxy. See flag '1' for its meaning.
          + 3 - behave like flag 1 is set for a request and like flag 2 is
            set for a reply.
          + a - flags that UA from which message is received doesn't
            support symmetric RTP. (automatically sets the 'r' flag)
          + b - append branch specific variable to Call-ID when sending
            command to rtpproxy. This creates one rtpproxy session per
            unique variable. Works similar to the 1, 2 and 3 parameter,
            but is usefull when forking to multiple destinations on
            different address families or network segments, requiring
            different rtpproxy parameters. The variable value is taken
            from the “extra_id_pv”. When used, it must be used in every
            call to rtpproxy_manage(), rtpproxy_offer(), rtpproxy_answer()
            and rtpproxy_destroy() with the same contents of the PV. The b
            parameter may not be used in conjunction with the 1, 2 or 3
            parameter to use the Via branch in the Call-ID.
          + l - force “lookup”, that is, only rewrite SDP when
            corresponding session already exists in the RTP proxy. By
            default is on when the session is to be completed.
          + i, e - these flags specify the direction of the SIP message.
            These flags only make sense when rtpproxy is running in bridge
            mode. 'i' means internal network (LAN), 'e' means external
            network (WAN). 'i' corresponds to rtpproxy's first interface,
            'e' corresponds to rtpproxy's second interface. You always
            have to specify two flags to define the incoming network and
            the outgoing network. For example, 'ie' should be used for SIP
            message received from the local interface and sent out on the
            external interface, and 'ei' vice versa. Other options are
            'ii' and 'ee'. So, for example if a SIP requests is processed
            with 'ie' flags, the corresponding response must be processed
            with 'ie' flags.
            Note: As rtpproxy in bridge mode s per default asymmetric, you
            have to specify the 'w' flag for clients behind NAT! See also
            above notes!
          + x - this flag a shortcut for using the "ie" or "ei"-flags of
            RTP-Proxy, in order to do automatic bridging between IPv4 on
            the "internal network" and IPv6 on the "external network". The
            distinction is done by the given IP in the SDP, e.g. a IPv4
            Address will always call "ie" to the RTPProxy (IPv4(i) to
            IPv6(e)) and an IPv6Address will always call "ei" to the
            RTPProxy (IPv6(e) to IPv4(i)).
            Note: Please note, that this will only work properly with
            non-dual-stack user-agents or with dual-stack clients
            according to RFC6157 (which suggest ICE for Dual-Stack
            implementations). This short-cut will not work properly with
            RFC4091 (ANAT) compatible clients, which suggests having
            different m-lines with different IP-protocols grouped
            together.
          + f - instructs rtpproxy to ignore marks inserted by another
            rtpproxy in transit to indicate that the session is already
            goes through another proxy. Allows creating a chain of
            proxies.
          + r - flags that IP address in SDP should be trusted. Without
            this flag, rtpproxy ignores address in the SDP and uses source
            address of the SIP message as media address which is passed to
            the RTP proxy.
          + o - flags that IP from the origin description (o=) should be
            also changed.
          + c - flags to change the session-level SDP connection (c=) IP
            if media-description also includes connection information.
          + w - flags that for the UA from which message is received,
            support symmetric RTP must be forced.
          + zNN - requests the RTPproxy to perform re-packetization of RTP
            traffic coming from the UA which has sent the current message
            to increase or decrease payload size per each RTP packet
            forwarded if possible. The NN is the target payload size in
            ms, for the most codecs its value should be in 10ms
            increments, however for some codecs the increment could differ
            (e.g. 30ms for GSM or 20ms for G.723). The RTPproxy would
            select the closest value supported by the codec. This feature
            could be used for significantly reducing 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.

   This function can be used from ANY_ROUTE.

   Example 1.10. 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();
...
}

5.3.  rtpproxy_answer([flags [, ip_address]])

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

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

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

   Example 1.11. rtpproxy_answer usage

   See rtpproxy_offer() function example above for example.

5.4.  rtpproxy_destroy([flags])

   Tears down the RTPProxy session for the current call.

   This function can be used from ANY_ROUTE.

   Meaning of the parameters is as follows:
     * flags - flags to turn on some features.
          + 1 - append first Via branch to Call-ID when sending command to
            rtpproxy. This can be used to create one media session per
            branch on the rtpproxy. When sending a subsequent “delete”
            command to the rtpproxy, you can then stop just the session
            for a specific branch when passing the flag '1' or '2' in the
            “unforce_rtpproxy”, or stop all sessions for a call when not
            passing one of those two flags there. This is especially
            useful if you have serially forked call scenarios where
            rtpproxy gets an “update” command for a new branch, and then a
            “delete” command for the previous branch, which would
            otherwise delete the full call, breaking the subsequent
            “lookup” for the new branch. This flag is only supported by
            the ngcp-mediaproxy-ng rtpproxy at the moment!
          + 2 - append second Via branch to Call-ID when sending command
            to rtpproxy. See flag '1' for its meaning.
          + b - append branch specific variable to Call-ID when sending
            command to rtpproxy. See rtpproxy_offer() for details.
            <listitem>
            </listitem>
            t - do not include To tag to “delete” command to rtpproxy thus
            causing full call to be deleted. Useful for deleting unused
            rtpproxy call when 200 OK is received on a branch, where
            rtpproxy is not needed.

   Example 1.12. rtpproxy_destroy usage
...
rtpproxy_destroy();
...

5.5.  unforce_rtp_proxy()

   Same as rtpproxy_destroy().

5.6.  rtpproxy_manage([flags [, ip_address]])

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

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

   Functionality:
     * If INVITE with SDP, then do rtpproxy_offer()
     * If INVITE with SDP, when the tm module is loaded, mark transaction
       with internal flag FL_SDP_BODY to know that the 1xx and 2xx are for
       rtpproxy_answer()
     * If ACK with SDP, then do rtpproxy_answer()
     * If BYE or CANCEL, or called within a FAILURE_ROUTE[], then do
       unforce_rtpproxy()
     * If reply to INVITE with code >= 300 do unforce_rtpproxy()
     * If reply with SDP to INVITE having code 1xx and 2xx, then do
       rtpproxy_answer() if the request had SDP or tm is not loaded,
       otherwise do rtpproxy_offer()

   This function can be used from ANY_ROUTE.

   Example 1.13. rtpproxy_manage usage
...
rtpproxy_manage();
...

5.7.  rtpproxy_stream2uac(prompt_name, count),

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

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

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

   This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE.

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

   Example 1.14. rtpproxy_stream2xxx usage
...
    if (is_method("INVITE")) {
        rtpproxy_offer();
        if (detect_hold()) {
            rtpproxy_stream2uas("/var/rtpproxy/prompts/music_on_hold", "-1");
        } else {
            rtpproxy_stop_stream2uas();
        };
    };
...

5.8.  rtpproxy_stream2uas(prompt_name, count)

   See function rtpproxy_stream2uac(prompt_name, count).

5.9.  rtpproxy_stop_stream2uac(),

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

   These functions can be used from REQUEST_ROUTE, ONREPLY_ROUTE.

5.10.  start_recording()

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

   This function can be used from REQUEST_ROUTE and ONREPLY_ROUTE.

   Example 1.15. start_recording usage
...
start_recording();
...

5.11.  rtpproxy_stop_stream2uas(prompt_name, count)

   See function rtpproxy_stop_stream2uac(prompt_name, count).

6. Exported Pseudo Variables

   6.1. $rtpstart

6.1. $rtpstart

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

   Example 1.16. $rtpstat-Usage
...
    append_hf("X-RTP-Statistics: $rtpstat\r\n");
...

7. MI Commands

   7.1. nh_enable_rtpp
   7.2. nh_show_rtpp

7.1. nh_enable_rtpp

   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 of its instances will be enables/disabled.

   Example 1.17.  nh_enable_rtpp usage
...
$ kamctl fifo nh_enable_rtpp udp:192.168.2.133:8081 0
...

7.2. nh_show_rtpp

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

   No parameter.

   Example 1.18.  nh_show_rtpp usage
...
$ kamctl fifo nh_show_rtpp
...

Chapter 2. Frequently Asked Questions

   2.1. What happend with “rtpproxy_disable” parameter?
   2.2. Where can I find more about Kamailio?
   2.3. Where can I post a question about this module?
   2.4. How can I report a bug?

   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 Kamailio?

   Take a look at http://www.kamailio.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.sip-router.org/cgi-bin/mailman/listinfo/sr-users
     * Developer Mailing List -
       http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

   E-mails regarding any stable Kamailio release should be sent to
   <sr-users@lists.sip-router.org> and e-mails regarding development
   versions should be sent to <sr-dev@lists.sip-router.org>.

   If you want to keep the mail private, send it to
   <sr-users@lists.sip-router.org>.

   2.4.

   How can I report a bug?

   Please follow the guidelines provided at:
   http://sip-router.org/tracker.
