/
rtpengine_admin.xml
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rtpengine_admin.xml
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<!-- Module User's Guide -->
<chapter>
<title>&adminguide;</title>
<section id="overview" xreflabel="Overview">
<title>Overview</title>
<para>
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
<ulink url="https://github.com/sipwise/rtpengine"></ulink>.
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.
</para>
</section>
<section>
<title>Multiple &rtp; proxy usage</title>
<para>
The rtpengine module can support multiple &rtp; proxies for
balancing/distribution and control/selection purposes.
</para>
<para>
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
<quote><xref linkend="param_rtpengine_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 &rtp; proxy from the set.
</para>
<para>
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.
</para>
<para>
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.
</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
rtpengine_delete(), rtpengine_offer() or rtpengine_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 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.
</para>
</section>
<section id="dependencies" xreflabel="Dependencies">
<title>Dependencies</title>
<section>
<title>&osips; 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 rtpengine_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 &osips; with this module loaded:
<itemizedlist>
<listitem>
<para>
<emphasis>None</emphasis>.
</para>
</listitem>
</itemizedlist>
</para>
</section>
</section>
<section>
<title>Exported Parameters</title>
<section id="param_rtpengine_sock" xreflabel="rtpengine_sock">
<title><varname>rtpengine_sock</varname> (string)</title>
<para>
Definition of socket(s) used to connect to (a set) &rtp; proxy. 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>rtpengine_sock</varname> parameter</title>
<programlisting format="linespecific">
...
# 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")
...
</programlisting>
</example>
</section>
<section id="param_rtpengine_disable_tout" xreflabel="rtpengine_disable_tout">
<title><varname>rtpengine_disable_tout</varname> (integer)</title>
<para>
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.
</para>
<para>
<emphasis>
Default value is <quote>60</quote>.
</emphasis>
</para>
<example>
<title>Set <varname>rtpengine_disable_tout</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpengine", "rtpengine_disable_tout", 20)
...
</programlisting>
</example>
</section>
<section id="param_rtpengine_tout" xreflabel="rtpengine_tout">
<title><varname>rtpengine_tout</varname> (integer)</title>
<para>
Timeout value in waiting for reply from &rtp; proxy.
</para>
<para>
<emphasis>
Default value is <quote>1</quote>.
</emphasis>
</para>
<example>
<title>Set <varname>rtpengine_tout</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpengine", "rtpengine_tout", 2)
...
</programlisting>
</example>
</section>
<section id="param_rtpengine_retr" xreflabel="rtpengine_retr">
<title><varname>rtpengine_retr</varname> (integer)</title>
<para>
How many times the module 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>rtpengine_retr</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpengine", "rtpengine_retr", 2)
...
</programlisting>
</example>
</section>
<section id="param_extra_id_pv" xreflabel="extra_id_pv">
<title><varname>extra_id_pv</varname> (string)</title>
<para>
The parameter sets the PV defination to use when the <quote>b</quote>
parameter is used on rtpengine_delete(), rtpengine_offer(),
rtpengine_answer() or rtpengine_manage() command.
</para><para>
Default is empty, the <quote>b</quote> parameter may not be used then.
</para>
<example>
<title>Set <varname>extra_id_pv</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpengine", "extra_id_pv", "$avp(extra_id)")
...
</programlisting>
</example>
</section>
<section id="param_setid_avp" xreflabel="setid_avp">
<title><varname>setid_avp</varname> (string)</title>
<para>
The parameter defines an AVP that, if set,
determines which &rtp; proxy set
rtpengine_offer(), rtpengine_answer(),
rtpengine_delete(), and rtpengine_manage()
functions use.
</para>
<para>
There is no default value.
</para>
<example>
<title>Set <varname>setid_avp</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpengine", "setid_avp", "$avp(setid)")
...
</programlisting>
</example>
</section>
<section id="param_db_url" xreflabel="db_url">
<title><varname>db_url</varname> (string)</title>
<para>
Database URL, used to load RTPEngines sockets
from db, instead of specifying them in the
script (<xref linkend="param_rtpengine_sock"/>
module parameter).
</para>
<para>
Default value is <quote>NULL</quote>, no database
is used.
</para>
<example>
<title>Set <varname>db_url</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpengine", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
...
</programlisting>
</example>
</section>
<section id="param_db_table" xreflabel="db_table">
<title><varname>db_table</varname> (string)</title>
<para>
The table where the RTPEngines sockets are stored.
Used when Database URL is provisioned.
</para>
<para>
Default value is <quote>rtpengines</quote>.
</para>
<example>
<title>Set <varname>db_table</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpengine", "db_table", "rtpengine_new")
...
</programlisting>
</example>
</section>
<section id="param_socket_column" xreflabel="socket_column">
<title><varname>socket_column</varname> (string)</title>
<para>
The name of the rtpengine socket column in the database table.
</para>
<para>
Default value is <quote>socket</quote>.
</para>
<example>
<title>Set <varname>socket_column</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpengine", "socket_column", "sock")
...
</programlisting>
</example>
</section>
<section id="param_set_column" xreflabel="set_column">
<title><varname>set_column</varname> (string)</title>
<para>
The name of the rtpengine set column in the database table.
</para>
<para>
Default value is <quote>set_id</quote>.
</para>
<example>
<title>Set <varname>set_column</varname> parameter</title>
<programlisting format="linespecific">
...
modparam("rtpengine", "set_column", "set_new")
...
</programlisting>
</example>
</section>
</section>
<section>
<title>Exported Functions</title>
<section id="func_rtpengine_use_set" xreflabel="rtpengine_use_set()">
<title>
<function moreinfo="none">rtpengine_use_set(setid)</function>
</title>
<para>
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.
</para>
<para>
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
BRANCH_ROUTE.
</para>
<example>
<title><function>rtpengine_use_set</function> usage</title>
<programlisting format="linespecific">
...
rtpengine_use_set(2);
rtpengine_offer();
...
</programlisting>
</example>
</section>
<section id="func_rtpengine_offer" xreflabel="rtpengine_offer()">
<title>
<function moreinfo="none">rtpengine_offer([flags[, sock_var[, sdp_pvar[, body]]]])</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(string, optional)</emphasis> - flags to turn on some features.
</para>
<para>The <quote>flags</quote> string is a list of space-separated items. Each item
is either an individual token, or a token in <quote>key=value</quote> format. The
possible tokens are described below.</para>
<para> When passing an option that &osips; is not aware of, it will be
blindly sent to the rtpengine daemon to be processed.</para>
<itemizedlist>
<listitem><para>
<emphasis>via-branch=...</emphasis> - Include the <quote>branch</quote>
value of one of the <quote>Via</quote> headers in the request to the
&rtp; proxy. Possible values are:
<quote>1</quote> - use the first <quote>Via</quote> header;
<quote>2</quote> - use the second <quote>Via</quote> header;
<quote>auto</quote> - use the first <quote>Via</quote> header if this is
a request, or the second one if this is a reply;
<quote>extra</quote> - don't take the value from a header, but instead use
the value of the <quote><xref linkend="param_extra_id_pv"/></quote> variable.
This can be used to create one media session per branch
on the &rtp; proxy. When sending a subsequent <quote>delete</quote> 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 <quote>rtpengine_delete</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 the &rtp; proxy
gets an <quote>offer</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>answer</quote> for the
new branch. <emphasis>This flag is only supported by the Sipwise rtpengine
&rtp; proxy at the moment!</emphasis>
</para></listitem>
<listitem><para>
<emphasis>asymmetric</emphasis> - flags that UA from which message is
received doesn't support symmetric RTP. (automatically sets the 'r' flag)
</para></listitem>
<listitem><para>
<emphasis>force-answer</emphasis> - force <quote>answer</quote>, 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.
</para></listitem>
<listitem><para>
<emphasis>in-iface=..., out-iface=...</emphasis> - these flags specify the direction
the SIP message. These flags only make sense when the &rtp; proxy is running
in bridge mode. <quote>in-iface</quote> should indicate the proxy's inbound
interface, and <quote>out-iface</quote> 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,
<quote>in-iface=internal out-iface=external</quote> should be
used for SIP message received from the local interface and sent out on the
external interface.
</para></listitem>
<listitem><para>
<emphasis>internal, external</emphasis> - these the old flags used to
specify the direction of call. They are now obsolate, being replaced by the
<quote>in-iface=internal out-iface=external</quote> configuration.
</para></listitem>
<listitem><para>
<emphasis>auto-bridge</emphasis> - this flag an alternative to the
<quote>internal</quote> and <quote>external</quote> 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.
</para></listitem>
<listitem><para>
<emphasis>address-family=...</emphasis> - instructs the &rtp; proxy that the
recipient of this &sdp; body expects to see addresses of a particular family.
Possible values are <quote>IP4</quote> and <quote>IP6</quote>. For example,
if the &sdp; body contains IPv4 addresses but the recipient only speaks IPv6,
you would use <quote>address-family=IP6</quote> to bridge between the two
address families.
</para><para>
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 <quote>offer</quote>,
but not in the <quote>answer</quote>.
</para><para>
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.
</para></listitem>
<listitem><para>
<emphasis>force</emphasis> - 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.
</para></listitem>
<listitem><para>
<emphasis>trust-address</emphasis> - 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.
</para></listitem>
<listitem><para>
<emphasis>replace-origin</emphasis> - flags that IP from the origin
description (o=) should be also changed.
</para></listitem>
<listitem><para>
<emphasis>replace-session-connection</emphasis> - flags to change the session-level
SDP connection (c=) IP if media description also includes
connection information.
</para></listitem>
<listitem><para>
<emphasis>symmetric</emphasis> - flags that for the UA from which
message is received, support symmetric RTP must be forced.
</para></listitem>
<listitem><para>
<emphasis>repacketize=NN</emphasis> - 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.
</para></listitem>
<listitem><para>
<emphasis>loop-protect</emphasis> - flag that instructs &rtp; to
avoid rewriting the SDP when looping the same message.
</para></listitem>
<listitem><para>
<emphasis>ICE=...</emphasis> - controls the &rtp; proxy's behaviour
regarding ICE attributes within the &sdp; body. Possible values
are: <quote>force</quote> -
discard any ICE attributes already present in the &sdp; body
and then generate and insert new ICE data, leaving itself
as the <emphasis>only</emphasis> ICE candidates;
<quote>remove</quote> instructs the &rtp; proxy to discard
any ICE attributes and not insert any new ones into the &sdp;.
The default (if no <quote>ICE=...</quote> 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
<emphasis>additional</emphasis> ICE candidate. Other
&sdp; substitutions (c=, m=, etc) are unaffected by this flag.
</para></listitem>
<listitem><para>
<emphasis>RTP, SRTP, AVP, AVPF</emphasis> - 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 <quote>SRTP</quote> flag indicates that
SRTP should be used, while <quote>RTP</quote> indicates that SRTP should not be used.
<quote>AVPF</quote> indicates that the advanced RTCP profile with feedback messages
should be used, and <quote>AVP</quote> indicates that the regular RTCP profile
should be used. See also the next set of flags below.
</para></listitem>
<listitem><para>
<emphasis>RTP/AVP, RTP/SAVP, RTP/AVPF, RTP/SAVPF</emphasis> - 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
<quote>RTP/SAVPF</quote> has the same effect as giving the two flags
<quote>SRTP AVPF</quote>.
</para></listitem>
<listitem><para>
<emphasis>to-tag</emphasis> - force inclusion of the <quote>To</quote> tag.
Normally, the <quote>To</quote> tag is always included when present, except
for <quote>delete</quote> messages. Including the <quote>To</quote> tag in
a <quote>delete</quote> messages allows you to be more selective about which
dialogues within a call are being torn down.
</para></listitem>
<listitem><para>
<emphasis>to-tag=...</emphasis> - use the specified string as <quote>To</quote>
tag instead of the actual <quote>To</quote> tag from the &sip; message, and
force inclusion of the tag in the message as per above.
</para></listitem>
<listitem><para>
<emphasis>from-tag=...</emphasis> - use the specified string as
<quote>From</quote> tag instead of the actual <quote>From</quote>
tag from the &sip; message.
</para></listitem>
<listitem><para>
<emphasis>call-id=...</emphasis> - use the specified string as
<quote>Call-ID</quote> instead of the actual <quote>Call-ID</quote>
from the &sip; message.
</para></listitem>
<listitem><para>
<emphasis>rtcp-mux-demux</emphasis> - if rtcp-mux (RFC 5761) was
offered, make the &rtp; proxy accept the offer, but not offer it to the
recipient of this message.
</para></listitem>
<listitem><para>
<emphasis>rtcp-mux-reject</emphasis> - if rtcp-mux was offered, make the
&rtp; proxy reject the offer, but still offer it to the recipient. Can be
combined with <quote>rtcp-mux-offer</quote> to always offer it.
</para></listitem>
<listitem><para>
<emphasis>rtcp-mux-offer</emphasis> - make the &rtp; proxy offer rtcp-mux
to the recipient of this message, regardless of whether it was offered
originally or not.
</para></listitem>
<listitem><para>
<emphasis>rtcp-mux-require</emphasis> - 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.
</para></listitem>
<listitem><para>
<emphasis>rtcp-mux-accept</emphasis> - 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 <quote>rtcp-mux-offer</quote> to always offer it.
</para></listitem>
<listitem><para>
<emphasis>media-address=...</emphasis> - force a particular media address to
be used in the &sdp; body. Address family is detected automatically.
</para></listitem>
<listitem><para>
<emphasis>record-call=yes/no</emphasis> - indicates whether rtpengine should
record the call or not. When using this parameter, you may pass further
information in the <quote>metadata</quote>.
</para></listitem>
<listitem><para>
<emphasis>transcode-CODEC</emphasis> - used only for offer, indicates that
rtpengine should transcode the CODEC towards the B-side. Example:
<emphasis>transcode-PCMA</emphasis> will present to the B-side the PCMA codec.
</para></listitem>
<listitem><para>
<emphasis>codec-strip-CODEC</emphasis> - used only for offer, indicates that
the A-side of the call will not end up talking CODEC. Example:
<emphasis>codec-strip-PCMA</emphasis> will prevent the A-side from receiving
the PCMA codec.
</para></listitem>
<listitem><para>
<emphasis>codec-mask-CODEC</emphasis> - used only for offer, indicates that
the A-side will use the CODEC, but it will not be presented to the B-side. Example:
<emphasis>codec-mask-PCMA</emphasis> will make the A-side receive the PCMA codec,
but B-side will use something else.
</para></listitem>
</itemizedlist>
</listitem>
<listitem><para>
<emphasis>sock_var(var, optional)</emphasis> - variable used to store the rtpengine
socket chosen for this call.
</para></listitem>
<listitem><para>
<emphasis>sdp_var(var, optional)</emphasis> - variable used to store the full SDP
received from rtpengine. You can perform any additional changes on this
string. <emphasis>Important: </emphasis> when providing this variable, the
message body is no longer changed, so you have to manually replace it!.
</para></listitem>
<listitem><para>
<emphasis>body(string, optional)</emphasis> - used to provide a specific body
to the rtpengine_* function. If this parameter is missing the body of
the current message is used.
</para></listitem>
</itemizedlist>
<para>
This function can be used from ANY_ROUTE.
</para>
<example>
<title><function>rtpengine_offer</function> usage</title>
<programlisting format="linespecific">
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();
...
}
</programlisting>
</example>
<example>
<title><function>rtpengine_offer</function> usage with body replace</title>
<programlisting format="linespecific">
...
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");
}
...
</programlisting>
</example>
<example>
<title><function>rtpengine_offer</function> usage with call recording</title>
<programlisting format="linespecific">
...
$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=" + $(json(recording_keys){s.encode.hexa});
rtpengine_offer($var(rtpengine_flags));
...
</programlisting>
</example>
<example>
<title><function>rtpengine_offer</function> usage for transcoding</title>
<programlisting format="linespecific">
...
# 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 ...");
...
</programlisting>
</example>
</section>
<section id="func_rtpengine_answer" xreflabel="rtpengine_answer()">
<title>
<function moreinfo="none">rtpengine_answer([flags[, sock_pvar[, sdp_pvar[, body]]]])</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 rtpengine_offer() 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>rtpengine_answer</function> usage</title>
<para>
See rtpengine_offer() function example above for examples.
</para>
</example>
</section>
<section id="func_rtpengine_delete" xreflabel="rtpengine_delete()">
<title>
<function moreinfo="none">rtpengine_delete([flags[, sock_var]])</function>
</title>
<para>
Tears down the RTPProxy session for the current call.
</para>
<para>
See rtpengine_offer() function description above for the meaning of the
parameters. Note that not all flags make sense for a <quote>delete</quote>.
</para>
<para>
This function can be used from ANY_ROUTE.
</para>
<example>
<title><function>rtpengine_delete</function> usage</title>
<programlisting format="linespecific">
...
rtpengine_delete();
...
</programlisting>
</example>
</section>
<section id="func_rtpengine_manage" xreflabel="rtpengine_manage()">
<title>
<function moreinfo="none">rtpengine_manage([flags[, sock_var[, sdp_var[, body]]]])</function>
</title>
<para>
Manage the RTPProxy 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.
</para>
<para>
It can take the same parameters as <function>rtpengine_offer().</function>
The flags parameter to rtpengine_manage() can be a configuration variable
containing the flags as a string.
</para>
<para>
Functionality:
</para>
<itemizedlist>
<listitem>
<para>
If INVITE with SDP, then do <function>rtpengine_offer()</function>
</para>
</listitem>
<listitem>
<para>
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
<function>rtpengine_answer()</function>
</para>
</listitem>
<listitem>
<para>
If ACK with SDP, then do <function>rtpengine_answer()</function>
</para>
</listitem>
<listitem>
<para>
If BYE or CANCEL, or called within a FAILURE_ROUTE[], then do <function>rtpengine_delete()</function>
</para>
</listitem>
<listitem>
<para>
If reply to INVITE with code >= 300 do <function>rtpengine_delete()</function>
</para>
</listitem>
<listitem>
<para>
If reply with SDP to INVITE having code 1xx and 2xx, then
do <function>rtpengine_answer()</function> if the request had SDP or tm is not loaded,
otherwise do <function>rtpengine_offer()</function>
</para>
</listitem>
</itemizedlist>
<para>
This function can be used from ANY_ROUTE.
</para>
<example>
<title><function>rtpengine_manage</function> usage</title>
<programlisting format="linespecific">
...
rtpengine_manage();
...
</programlisting>
</example>
</section>
<section id="func_rtpengine_start_recording" xreflabel="rtpengine_start_recording()">
<title>
<function moreinfo="none">rtpengine_start_recording([flags [, sock_var]])</function>
</title>
<para>
This function will send a signal to the &rtp; proxy to record
the RTP stream on the &rtp; proxy.
</para>
<para>Meaning of the parameters is as follows:</para>
<itemizedlist>
<listitem><para>
<emphasis>flags(string, optional)</emphasis> - flags used to change the behavior
of the recorder. An importat value to set is the <emphasis>call-id</emphasis>
value, which can be used to start recording a different call than the requested one.
</para></listitem>
<listitem><para>
<emphasis>sock_var(var, optional)</emphasis> - variable used to store the rtpengine
socket chosen for this call.
</para></listitem>
</itemizedlist>
<para>
This function can be used from any route.
</para>
<example>
<title><function>rtpengine_start_recording</function> usage</title>
<programlisting format="linespecific">
...
rtpengine_start_recording();
...
</programlisting>
</example>
</section>
<section id="func_rtpengine_stop_recording" xreflabel="rtpengine_stop_recording()">
<title>
<function moreinfo="none">rtpengine_stop_recording([flags [, sock_var]])</function>
</title>
<para>
This function will send a signal to the &rtp; proxy to stop
recording the RTP stream on the &rtp; proxy.
</para>
<para>Meaning of the parameters is as follows:</para>
<itemizedlist>
<listitem><para>
<emphasis>flags(string, optional)</emphasis> - flags used to change the behavior
of the recorder. An importat value to set is the <emphasis>call-id</emphasis>
value, which can be used to start recording a different call than the requested one.
</para></listitem>
<listitem><para>
<emphasis>sock_var(var, optional)</emphasis> - variable used to store the rtpengine
socket chosen for this call.
</para></listitem>
</itemizedlist>
<para>
This function can be used from any route.
</para>
<example>
<title><function>rtpengine_stop_recording</function> usage</title>
<programlisting format="linespecific">
...
rtpengine_stop_recording();
...
</programlisting>
</example>
</section>
<section id="func_rtpengine_play_media" xreflabel="rtpengine_play_media()">
<title>
<function moreinfo="none">rtpengine_play_media(flags, [duration_spec[, sock_var[, sockvar]]])</function>
</title>
<para>
This function will start playing a media file to one of the endpoints.
</para>
<para>Meaning of the parameters is as follows:</para>
<itemizedlist>
<listitem><para>
<emphasis>flags(string)</emphasis> - a list of flags simialar to
the other functions. One of the <emphasis>file</emphasis>,
<emphasis>blob</emphasis> or <emphasis>db-id</emphasis> parameters
is mandatory to indicate the content of the media file to be played.
<emphasis>file</emphasis> is a common choice for specifying rtpengine
to get media from a file path, <emphasis>blob</emphasis> to take
the content from an inline string and <emphasis>db-id</emphasis> to
get the content from the database.
<para>
The direction of the media stream is controlled by the
<emphasis>from-tag</emphasis> parameter, <emphasis>address</emphasis>
(media address from the SDP), or <emphasis>label</emphasis>, 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.
</para>
</para></listitem>
<listitem><para>
<emphasis>duration_spec(var, optional)</emphasis> - a pseudo variable that
will contain the duration of the played file. It will be set to
<emphasis>-1</emphasis> if the duration could not be determined.
</para></listitem>
<listitem><para>
<emphasis>sock_var(var, optional)</emphasis> - variable used to store the rtpengine
socket chosen for this call.
</para></listitem>
</itemizedlist>
<para>
This function can be used from any route.
</para>
<example>
<title>Ringback tone using <function>rtpengine_play_media</function></title>
<programlisting format="linespecific">
...
if (is_method("INVITE") && !has_totag())
rtpengine_play_media("file=/path/to/ringback_tone_file.wav");
...
</programlisting>
</example>
<example>
<title>Manage music on hold using <function>rtpengine_play_media</function></title>
<programlisting format="linespecific">
...
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_file.wav");
} else if ($dlg_val(on_hold) == "1") {
$dlg_val(on_hold) = "0";
rtpengine_stop_media("from-tag=$tt");
}
}
...
</programlisting>
</example>
</section>
<section id="func_rtpengine_stop_media" xreflabel="rtpengine_stop_media()">
<title>
<function moreinfo="none">rtpengine_stop_media([flags[, sockvar]])</function>
</title>
<para>
This function will stop playing a media file previously started
by a <function>rtpengine_play_media()</function> 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
<function>rtpengine_play_media()</function> call, otherwise
RTPEngine will not be able to stop media playing.
</para>
<para>
This function can be used from any route.
</para>
<example>
<title>Ringback tone stop using <function>rtpengine_stop_media</function></title>
<programlisting format="linespecific">
...
if (is_method("INVITE") && $rs == 200)
rtpengine_stop_media();
...
</programlisting>
</example>
</section>
<section id="func_rtpengine_block_media" xreflabel="rtpengine_block_media()">
<title>
<function moreinfo="none">rtpengine_stop_media([flags[, sockvar]])</function>
</title>
<para>
This function will block the media sent from one of the endpoints.
The direction to be blocked is controled by the <emphasis>flags</emphasis>
parameter, the <emphasis>from-tag</emphasis> value.
</para>
<para>
This function can be used from any route.
</para>
<example>
<title>Example of <function>rtpengine_block_media</function> usage</title>
<programlisting format="linespecific">
...
rtpengine_block_media();
...
</programlisting>
</example>
</section>
<section id="func_rtpengine_unblock_media" xreflabel="rtpengine_unblock_media()">
<title>
<function moreinfo="none">rtpengine_unstop_media([flags[, sockvar]])</function>
</title>
<para>
This function will resume/unblock the media sent from one of the endpoints.
The direction to be blocked is controled by the <emphasis>flags</emphasis>
parameter, the <emphasis>from-tag</emphasis> value.
</para>
<para>
This function can be used from any route.
</para>
<example>
<title>Example of <function>rtpengine_unblock_media</function> usage</title>
<programlisting format="linespecific">
...
rtpengine_unblock_media();
...