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kamailio/modules/rtpproxy/doc/rtpproxy_admin.xml

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<?xml version="1.0" encoding='ISO-8859-1'?>
<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook XML V4.4//EN"
"http://www.oasis-open.org/docbook/xml/4.4/docbookx.dtd" [
<!-- Include general documentation entities -->
<!ENTITY % docentities SYSTEM "../../../docbook/entities.xml">
%docentities;
]>
<!-- Module User's Guide -->
<chapter>
<title>&adminguide;</title>
<section>
<title>Overview</title>
<para>
This is a module that enables media streams to be proxied
via an rtpproxy.
</para>
<para>
Known devices that get along over &nat;s with rtpproxy are ATAs
(as clients) and Cisco Gateways (since 12.2(T)) as servers. See <ulink
url="http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_guide09186a0080110bf9.html">
http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_guide09186a0080110bf9.html"></ulink>
</para>
</section>
<section>
<title>Multiple RTPProxy usage</title>
<para>
Currently, the rtpproxy module can support multiple rtpproxies for
balancing/distribution and control/selection purposes.
</para>
<para>
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
<quote>rtpproxy_sock</quote> module parameter definition for syntax
description.
</para>
<para>
The balancing inside a set is done automatically by the module based on
the weight of each rtpproxy from the set.
</para>
<para>
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.
</para>
<para>
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.
</para>
<para>
IMPORTANT: if you use multiple sets, take care and use the same set for
both rtpproxy_offer()/rtpproxy_answer() and unforce_rtpproxy()!!
</para>
</section>
<section>
<title>Dependencies</title>
<section>
<title>&kamailio; Modules</title>
<para>
The following modules must be loaded before this module:
<itemizedlist>
<listitem>
<para>
<emphasis>tm module</emphasis> - (optional) if you want to
have rtpproxy_manage() fully functional
</para>
</listitem>
</itemizedlist>
</para>
</section>
<section>
<title>External Libraries or Applications</title>
<para>
The following libraries or applications must be installed before
running &kamailio; with this module loaded:
<itemizedlist>
<listitem>
<para>
<emphasis>None</emphasis>.
</para>
</listitem>
</itemizedlist>
</para>
</section>
</section>
<section>
<title>Parameters</title>
<section>
<title><varname>rtpproxy_sock</varname> (string)</title>
<para>
Definition of socket(s) used to connect to (a set) RTPProxy. It may
specify a UNIX socket or an IPv4/IPv6 UDP socket.
</para>
<para>
<emphasis>
Default value is <quote>NONE</quote> (disabled).
</emphasis>
</para>
<example>
<title>Set <varname>rtpproxy_sock</varname> parameter</title>
<programlisting format="linespecific">
...
# 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")
...
</programlisting>
</example>
</section>
<section>
<title><varname>rtpproxy_disable_tout</varname> (integer)</title>
<para>
Once RTPProxy was found unreachable and marked as disable, rtpproxy
will not attempt to establish communication to RTPProxy for
rtpproxy_disable_tout seconds.
</para>
<para>
<emphasis>
Default value is <quote>60</quote>.
</emphasis>
</para>
<example>
<title>Set <varname>rtpproxy_disable_tout</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpproxy", "rtpproxy_disable_tout", 20)
...
</programlisting>
</example>
</section>
<section>
<title><varname>rtpproxy_tout</varname> (integer)</title>
<para>
Timeout value in waiting for reply from RTPProxy.
</para>
<para>
<emphasis>
Default value is <quote>1</quote>.
</emphasis>
</para>
<example>
<title>Set <varname>rtpproxy_tout</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpproxy", "rtpproxy_tout", 2)
...
</programlisting>
</example>
</section>
<section>
<title><varname>rtpproxy_retr</varname> (integer)</title>
<para>
How many times rtpproxy should retry to send and receive after
timeout was generated.
</para>
<para>
<emphasis>
Default value is <quote>5</quote>.
</emphasis>
</para>
<example>
<title>Set <varname>rtpproxy_retr</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpproxy", "rtpproxy_retr", 2)
...
</programlisting>
</example>
</section>
<section>
<title><varname>force_socket</varname> (string)</title>
<para>
Socket to be forced in communicating to RTPProxy. It makes sense only
for UDP communication. If no one specified, the OS will choose.
</para>
<para>
<emphasis>
Default value is <quote>NULL</quote>.
</emphasis>
</para>
<example>
<title>Set <varname>force_socket</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpproxy", "force_socket", "localhost:33333")
...
</programlisting>
</example>
</section>
<section>
<title><varname>nortpproxy_str</varname> (string)</title>
<para>
The parameter sets the SDP attribute used by rtpproxy to mark
the packet SDP informations have already been mangled.
</para>
<para>
If empty string, no marker will be added or checked.
</para>
<note><para>
The string must be a complete SDP line, including the EOH (\r\n).
</para></note>
<para>
<emphasis>
Default value is <quote>a=nortpproxy:yes\r\n</quote>.
</emphasis>
</para>
<example>
<title>Set <varname>nortpproxy_str</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpproxy", "nortpproxy_str", "a=sdpmangled:yes\r\n")
...
</programlisting>
</example>
</section>
<section>
<title><varname>timeout_socket</varname> (string)</title>
<para>
The parameter sets timeout socket, which is transmitted to the RTP-Proxy.
</para>
<para>
If it is an empty string, no timeout socket will be transmitted to the RTP-Proxy.
</para>
<para>
<emphasis>
Default value is <quote></quote> (nothing).
</emphasis>
</para>
<example>
<title>Set <varname>timeout_socket</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("nathelper", "timeout_socket", "xmlrpc:http://127.0.0.1:8000/RPC2")
...
</programlisting>
</example>
</section>
</section>
<section>
<title>Functions</title>
<section>
<title>
<function moreinfo="none">set_rtp_proxy_set()</function>
</title>
<para>
Sets the Id of the rtpproxy set to be used for the next
unforce_rtp_proxy(), rtpproxy_offer() or rtpproxy_answer()
command.
</para>
<para>
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
BRANCH_ROUTE.
</para>
<example>
<title><function>fix_nated_contact</function> usage</title>
<programlisting format="linespecific">
...
set_rtp_proxy_set("2");
rtpproxy_offer();
...
</programlisting>
</example>
</section>
<section>
<title>
<function moreinfo="none">rtpproxy_offer([flags [, ip_address]])</function>
</title>
<para>
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.
</para>
<para>Meaning of the parameters is as follows:</para>
<itemizedlist>
<listitem>
<para>
<emphasis>flags</emphasis> - flags to turn on some features.
</para>
<itemizedlist>
<listitem><para>
<emphasis>1</emphasis> - 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 <quote>delete</quote> command to
the rtpproxy, you can then stop just the session for a specific branch when
passing the flag '1' or '2' in the <quote>unforce_rtpproxy</quote>, 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 <quote>update</quote> command for a new branch, and then a
<quote>delete</quote> command for the previous branch, which would otherwise
delete the full call, breaking the subsequent <quote>lookup</quote> for the
new branch. <emphasis>This flag is only supported by the ngcp-mediaproxy-ng
rtpproxy at the moment!</emphasis>
</para></listitem>
<listitem><para>
<emphasis>2</emphasis> - append second Via branch to Call-ID when sending
command to rtpproxy. See flag '1' for its meaning.
</para></listitem>
<listitem><para>
<emphasis>a</emphasis> - flags that UA from which message is
received doesn't support symmetric RTP. (automatically sets the 'r' flag)
</para></listitem>
<listitem><para>
<emphasis>l</emphasis> - force <quote>lookup</quote>, 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.
</para></listitem>
<listitem><para>
<emphasis>i, e</emphasis> - 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.
</para><para>
Note: As rtpproxy is in bridge mode per default asymmetric, you have to specify
the 'w' flag for clients behind NAT! See also above notes!
</para></listitem>
<listitem><para>
<emphasis>f</emphasis> - 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.
</para></listitem>
<listitem><para>
<emphasis>r</emphasis> - 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.
</para></listitem>
<listitem><para>
<emphasis>o</emphasis> - flags that IP from the origin
description (o=) should be also changed.
</para></listitem>
<listitem><para>
<emphasis>c</emphasis> - flags to change the session-level
SDP connection (c=) IP if media-description also includes
connection information.
</para></listitem>
<listitem><para>
<emphasis>w</emphasis> - flags that for the UA from which
message is received, support symmetric RTP must be forced.
</para></listitem>
<listitem><para>
<emphasis>zNN</emphasis> - 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.
</para></listitem>
</itemizedlist>
</listitem>
<listitem><para>
<emphasis>ip_address</emphasis> - new SDP IP address.
</para></listitem>
</itemizedlist>
<para>
This function can be used from ANY_ROUTE.
</para>
<example>
<title><function>rtpproxy_offer</function> usage</title>
<programlisting format="linespecific">
route {
...
if (is_method("INVITE")) {
if (has_sdp()) {
if (rtpproxy_offer())
t_on_reply("1");
} else {
t_on_reply("2");
}
}
if (is_method("ACK") &amp;&amp; has_sdp())
rtpproxy_answer();
...
}
onreply_route[1]
{
...
if (has_sdp())
rtpproxy_answer();
...
}
onreply_route[2]
{
...
if (has_sdp())
rtpproxy_offer();
...
}
</programlisting>
</example>
</section>
<section>
<title>
<function moreinfo="none">rtpproxy_answer([flags [, ip_address]])</function>
</title>
<para>
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.
</para>
<para>
See rtpproxy_answer() function description above for the meaning of the
parameters.
</para>
<para>
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
FAILURE_ROUTE, BRANCH_ROUTE.
</para>
<example>
<title><function>rtpproxy_answer</function> usage</title>
<para>
See rtpproxy_offer() function example above for example.
</para>
</example>
</section>
<section>
<title>
<function moreinfo="none">rtpproxy_destroy([flags])</function>
</title>
<para>
Tears down the RTPProxy session for the current call.
</para>
<para>
This function can be used from ANY_ROUTE.
</para>
<para>Meaning of the parameters is as follows:</para>
<itemizedlist>
<listitem>
<para>
<emphasis>flags</emphasis> - flags to turn on some features.
</para>
<itemizedlist>
<listitem><para>
<emphasis>1</emphasis> - 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 <quote>delete</quote> command to
the rtpproxy, you can then stop just the session for a specific branch when
passing the flag '1' or '2' in the <quote>unforce_rtpproxy</quote>, 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 <quote>update</quote> command for a new branch, and then a
<quote>delete</quote> command for the previous branch, which would otherwise
delete the full call, breaking the subsequent <quote>lookup</quote> for the
new branch. <emphasis>This flag is only supported by the ngcp-mediaproxy-ng
rtpproxy at the moment!</emphasis>
</para></listitem>
<listitem><para>
<emphasis>2</emphasis> - append second Via branch to Call-ID when sending
command to rtpproxy. See flag '1' for its meaning.
</para></listitem>
</itemizedlist>
</listitem>
</itemizedlist>
<example>
<title><function>rtpproxy_destroy</function> usage</title>
<programlisting format="linespecific">
...
rtpproxy_destroy();
...
</programlisting>
</example>
</section>
<section>
<title>
<function moreinfo="none">unforce_rtp_proxy()</function>
</title>
<para>
Same as rtpproxy_destroy().
</para>
</section>
<section>
<title>
<function moreinfo="none">rtpproxy_manage([flags [, ip_address]])</function>
</title>
<para>
Manage the RTPProxy session - it combines the functionality of
rtpproxy_offer(), rtpproxy_answer() and unfroce_rtpproxy(), detecting
internally based on message type and metod which one to execute.
</para>
<para>
It can take same kind of parameters as rtpproxy_offer().
</para>
<para>
Functinality:
</para>
<itemizedlist>
<listitem>
<para>
if INVITE with SDP, then do rtpproxy offer
</para>
</listitem>
<listitem>
<para>
if INVITE with SDP, when tm is loaded, mark transaction with
internal flag FL_SDP_BODY to know that the 1xx and 2xx are for
rtpproxy answer
</para>
</listitem>
<listitem>
<para>
if ACK with SDP, then do rtpproxy answer
</para>
</listitem>
<listitem>
<para>
if BYE or CANCEL, or called within a failure_route[], then do unforce rtpproxy
</para>
</listitem>
<listitem>
<para>
if reply to INVITE with code >= 300 do unfrce rtp proxy
</para>
</listitem>
<listitem>
<para>
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
</para>
</listitem>
</itemizedlist>
<para>
This function can be used from ANY_ROUTE.
</para>
<example>
<title><function>rtpproxy_manage</function> usage</title>
<programlisting format="linespecific">
...
rtpproxy_manage();
...
</programlisting>
</example>
</section>
<section id="rtpproxy_stream2uac">
<title>
<function>rtpproxy_stream2uac(prompt_name, count)</function>,
</title>
<para>
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
<function>rtpproxy_stream2uac</function> in the request processing
block on ACK transaction will play the prompt to the UA that has
generated original INVITE and ACK while
<function>rtpproxy_stop_stream2uas</function> on 183 in reply
processing block will play the prompt to the UA that has generated 183.
</para>
<para>
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 <function>rtpproxy_stop_stream2xxx</function> is issued.
</para>
<para>
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 <function>rtpproxy_offer</function>
or <function>rtpproxy_answer</function>.
</para>
<para>
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE.
</para>
<para>Meaning of the parameters is as follows:</para>
<itemizedlist>
<listitem>
<para>
<emphasis>prompt_name</emphasis> - name of the prompt to
stream. Should be either absolute pathname or pathname
relative to the directory where RTPproxy runs.
</para>
</listitem>
<listitem>
<para>
<emphasis>count</emphasis> - number of times the prompt
should be repeated. The value of -1 means that it will
be streaming in loop indefinitely, until appropriate
<function>rtpproxy_stop_stream2xxx</function> is issued.
</para>
</listitem>
</itemizedlist>
<example>
<title><function>rtpproxy_stream2xxx</function> usage</title>
<programlisting>
...
if (is_method("INVITE")) {
rtpproxy_offer();
if (detect_hold()) {
rtpproxy_stream2uas("/var/rtpproxy/prompts/music_on_hold", "-1");
} else {
rtpproxy_stop_stream2uas();
};
};
...
</programlisting>
</example>
</section>
<section id="rtpproxy_stream2uas">
<title>
<function>rtpproxy_stream2uas(prompt_name, count)</function>
</title>
<para>
See function <function>rtpproxy_stream2uac(prompt_name, count)</function>.
</para>
</section>
<section id="rtpproxy_stop_stream2uac">
<title>
<function>rtpproxy_stop_stream2uac()</function>,
</title>
<para>
Stop streaming of announcement/prompt/MOH started previously by the
respective <function>rtpproxy_stream2xxx</function>. The uac/uas
suffix selects whose announcement relatively to tha current
transaction should be stopped - UAC or UAS.
</para>
<para>
These functions can be used from REQUEST_ROUTE, ONREPLY_ROUTE.
</para>
</section>
<section>
<title>
<function moreinfo="none">start_recording()</function>
</title>
<para>
This command will send a signal to the RTP-Proxy to record
the RTP stream on the RTP-Proxy.
</para>
<para>
This function can be used from REQUEST_ROUTE and ONREPLY_ROUTE.
</para>
<example>
<title><function>start_recording</function> usage</title>
<programlisting format="linespecific">
...
start_recording();
...
</programlisting>
</example>
</section>
<section id="rtpproxy_stop_stream2uas">
<title>
<function>rtpproxy_stop_stream2uas(prompt_name, count)</function>
</title>
<para>
See function <function>rtpproxy_stop_stream2uac(prompt_name, count)</function>.
</para>
</section>
</section>
<section>
<title>Exported Pseudo Variables</title>
<section>
<title><function moreinfo="none">$rtpstart</function></title>
<para>
Returns the RTP-Statistics from the RTP-Proxy. The RTP-Statistics from the RTP-Proxy
are provided as a string and it does contain several packet-counters. The statistics
must be retrieved before the session is deleted (before unforce_rtpproxy).
</para>
<example>
<title>$rtpstat-Usage</title>
<programlisting format="linespecific">
...
append_hf("X-RTP-Statistics: $rtpstat\r\n");
...
</programlisting>
</example>
</section>
</section>
<section>
<title><acronym>MI</acronym> Commands</title>
<section>
<title><function moreinfo="none">nh_enable_rtpp</function></title>
<para>
Enables a rtp proxy if parameter value is greater than 0.
Disables it if a zero value is given.
</para>
<para>
The first parameter is the rtp proxy url (exactly as defined in
the config file).
</para>
<para>
The second parameter value must be a number in decimal.
</para>
<para>
NOTE: if a rtpproxy is defined multiple times (in the same or
diferente sete), all its instances will be enables/disabled.
</para>
<example>
<title>
<function moreinfo="none">nh_enable_rtpp</function> usage</title>
<programlisting format="linespecific">
...
$ &ctltool; fifo nh_enable_rtpp udp:192.168.2.133:8081 0
...
</programlisting>
</example>
</section>
<section>
<title><function moreinfo="none">nh_show_rtpp</function></title>
<para>
Displays all the rtp proxies and their information: set and
status (disabled or not, weight and recheck_ticks).
</para>
<para>
No parameter.
</para>
<example>
<title>
<function moreinfo="none">nh_show_rtpp</function> usage</title>
<programlisting format="linespecific">
...
$ &ctltool; fifo nh_show_rtpp
...
</programlisting>
</example>
</section>
</section>
</chapter>