rtpproxy Module
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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_retr (integer)
1.5.6. default_set (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)
1.6. Exported Functions
1.6.1. rtpproxy_engage([[flags][, [ip_address][,
[set_id][, [sock_var][, ret_var]]]]])
1.6.2. rtpproxy_offer([[flags][, [ip_address][,
[set_id][, [sock_var][, [ret_var][,
[body_var]]]]]])
1.6.3. rtpproxy_answer([[flags][, [ip_address][,
[set_id][, [sock_var][, [ret_var][,
[body_var]]]]]]])
1.6.4. rtpproxy_unforce([[set_id][, sock_var]])
1.6.5. rtpproxy_stream2uac(prompt_name, count[,
[set_id][, sock_var]]),
rtpproxy_stream2uas(prompt_name, count[,
[set_id][, sock_var]])
1.6.6. rtpproxy_stop_stream2uac([[set_id][,
sock_var]]),
rtpproxy_stop_stream2uas([[set_id][,
sock_var]])
1.6.7. rtpproxy_start_recording([[set_id][,
[sock_var][, [flags][, [destination][,
mediastream]]]]])
1.6.8. rtpproxy_stats(up_pvar, down_var, sent_var,
fail_var[, [set_id][, sock_var]])
1.6.9. rtpproxy_all_stats(stats_avp[, [set_id][,
sock_var]])
1.7. Exported MI Functions
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
1.8.2. E_RTPPROXY_DTMF
2. Frequently Asked Questions
3. Contributors
3.1. By Commit Statistics
3.2. By Commit Activity
4. Documentation
4.1. Contributors
List of Tables
3.1. Top contributors by DevScore^(1), authored commits^(2) and
lines added/removed^(3)
3.2. Most recently active contributors^(1) to this module
List of Examples
1.1. Set 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 default_set parameter
1.7. Set nortpproxy_str parameter
1.8. Set db_url parameter
1.9. Set db_table parameter
1.10. Set rtpp_socket_col parameter
1.11. Set set_id parameter
1.12. Set rtpp_notify_socket parameter
1.13. rtpproxy_engage usage
1.14. rtpproxy_offer usage
1.15. rtpproxy_answer usage
1.16. rtpproxy_unforce usage
1.17. rtpproxy_stream2xxx usage
1.18. rtpproxy_start_recording usage
1.19. rtpproxy_stats usage
1.20. rtpproxy_all_stats usage
1.21. rtpproxy_enable usage
1.22. rtpproxy_show usage
1.23. 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.
Starting with OpenSIPS 2.1, engage_rtp_proxy(),
unforce_rtp_proxy() and start_recording() functions have been
fully replaced by rtpproxy_engage(), rtpproxy_unforce() and
rtpproxy_start_recording().
IMPORTANT: if you use multiple sets, make sure you use the same
set for both rtpproxy_offer()/rtpproxy_answer() and
rtpproxy_unforce()!!
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.
Note that RTPProxy version v2.0.0 has integrated this feature
upstream, therefore this patch is no longer needed.
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
rtpproxy_engage(), 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:
https://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 rtpproxy_engage 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, an IPv4/IPv6 UDP socket or an
IPv4/IPv6 TCP socket.
The definition also supports to specify a different IP that
will be advertised instead of the one returned by RTPProxy.
This is useful when having multiple RTPProxy servers that are
located behind NAT, and listen only on private intefaces, but
need to advertise a public one.
Default value is "NONE" (disabled).
Example 1.1. Set rtpproxy_sock parameter
...
# single rtproxy with specific weight
modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:22222=2")
# single rtpproxy with advertised address + weight
modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:22222|8.8.8.8=2")
# multiple rtproxies for LB
modparam("rtpproxy", "rtpproxy_sock",
"udp:localhost:22222 udp:localhost:22223 tcp:remote1:33422 tcp6:
remote2:32322")
# multiple sets of multiple rtproxies
modparam("rtpproxy", "rtpproxy_sock", "1 == udp:localhost:22222 udp:loca
lhost:22223")
modparam("rtpproxy", "rtpproxy_sock", "2 == udp:localhost:22223")
modparam("rtpproxy", "rtpproxy_sock", "2 == udp:localhost:22223|8.8.8.8"
)
...
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_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)
...
1.5.6. default_set (integer)
The parameter indicates the default RTPProxy set to be used
when provisioning an engine in the config file without an
explicit set, or when calling one of the rtpproxy_*() functions
without an explicit set.
Default value is set "0".
Example 1.6. Set default_set parameter
...
modparam("rtpproxy", "default_set", 1)
...
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 1.7. 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 1.8. 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 1.9. 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 1.10. 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 1.11. Set set_id parameter
...
modparam("rtpproxy", "set_id_col", "rtpp_set_id")
...
1.5.12. rtpp_notify_socket (string)
The socket OpenSIPS listens for notifications from RTPProxy.
Currently OpenSIPS can receive RTP timeout and DTMF events.
Default value is "NULL" - no notifications are received.
Example 1.12. Set rtpp_notify_socket parameter
...
modparam("rtpproxy", "rtpp_notify_socket", "tcp:10.10.10.10:9999")
# use an UNIX socket
modparam("rtpproxy", "rtpp_notify_socket", "unix:/tmp/rtpproxy.unix")
# or
modparam("rtpproxy", "rtpp_notify_socket", "/tmp/rtpproxy.unix")
...
1.6. Exported Functions
1.6.1. rtpproxy_engage([[flags][, [ip_address][, [set_id][,
[sock_var][, ret_var]]]]])
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(string, optional) - 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).
+ k - only create RTPProxy session, but do not modify
the SDP body. This is useful when you only want to
inject some media, but do not want to engage RTPProxy
in the entire call.
+ 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.
NOTE: when using RTPProxy in bridge mode, all sessions
are considered asymmetric (as oposed to symmetric if
used in normal mode). If you have symmetric clients
(this is the most common scenario), you'll have to
force the s!
+ 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[<SOCKET>] - flags that enables the notification
timeout for the session. One can specify an optional
"advertised" socket between the < and > tags. If the
socket is not specified, the value of
rtpp_notify_socket is used.
+ d[NNN] - enables DTMF notifications for this call. One
can optionally specify the payload type that DTMF will
be used for this call - it it is not specified,
RTPProxy uses the 101 pt. NOTE: this feature is
currently only available in the RTPProxy rtpp_2_1_dtmf
branch.
+ tNN - can be used to specify a RTP ttl for the caller.
The NN represents the timeout in seconds for that
stream. This can be useful in music on hold scenarios
where only one client is sending RTP.
+ TNN - Similar to the tNN paramaeter, but used for
tuning the calllee's ttl for RTP.
+ 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(string, optional) - new SDP IP address.
* set_id(int, optional) - the set used for this call.
* sock_var(var, optional) - variable used to store the
RTPProxy socket chosen for this call. Note that the
variable will only be populated in the initial request.
* ret_var(var, optional) - variable used to print the IP and
port the RTPProxy server is using for this call. This is
useful especially when using the rtp_cluster, which can
advertise multiple servers behind it. The format of the
value returned is IP:port. Note that the variable will only
be populated in the initial request.
This function can be used from REQUEST_ROUTE, FAILURE_ROUTE,
BRANCH_ROUTE.
Example 1.13. rtpproxy_engage usage
...
if (is_method("INVITE") && has_totag()) {
if ($var(setid) != 0) {
rtpproxy_engage(,,$var(setid), $var(proxy));
xlog("SCRIPT: RTPProxy server used is $var(proxy)\n");
} else {
rtpproxy_engage();
xlog("SCRIPT: using default RTPProxy set\n");
}
}
...
1.6.2. rtpproxy_offer([[flags][, [ip_address][, [set_id][,
[sock_var][, [ret_var][, [body_var]]]]]])
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.
The function receives the same parameters as rtpproxy_engage(),
as well as an extra parameter named body_var - this parameter
is used as an in-out variable for the body that should be used
to challenge RTP proxy server. If the variable is specified, it
is the function uses its content as the body to challenge, and
returns the resulted body in it. If not used, the message's
body is used, and the outgoing body is changed.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
FAILURE_ROUTE, BRANCH_ROUTE.
Example 1.14. rtpproxy_offer usage
route {
...
if (is_method("INVITE")) {
if (has_body("application/sdp")) {
if (rtpproxy_offer())
t_on_reply("1");
} else {
t_on_reply("2");
}
}
if (is_method("ACK") && has_body("application/sdp"))
rtpproxy_answer();
...
}
onreply_route[1]
{
...
if (has_body("application/sdp"))
rtpproxy_answer();
...
}
onreply_route[2]
{
...
if (has_body("application/sdp"))
rtpproxy_offer();
...
}
1.6.3. rtpproxy_answer([[flags][, [ip_address][, [set_id][,
[sock_var][, [ret_var][, [body_var]]]]]]])
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_offer() function description above for the meaning
of the parameters.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
FAILURE_ROUTE, BRANCH_ROUTE.
Example 1.15. rtpproxy_answer usage
See rtpproxy_offer() function example above for example.
1.6.4. rtpproxy_unforce([[set_id][, sock_var]])
Tears down the RTPProxy session for the current call.
Meaning of the parameters is as follows:
* set_id(int, optional) - the set used for this call.
* sock_var(var, optional) - variable used to store the
RTPProxy socket chosen for this call.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
FAILURE_ROUTE, BRANCH_ROUTE.
Example 1.16. rtpproxy_unforce usage
...
rtpproxy_unforce();
...
1.6.5. rtpproxy_stream2uac(prompt_name, count[, [set_id][,
sock_var]]), rtpproxy_stream2uas(prompt_name, count[, [set_id][,
sock_var]])
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 (string) - name of the prompt to stream. Should
be either absolute pathname or pathname relative to the
directory where RTPproxy runs.
* count (int) - 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.
* set_id(int, optional) - the set used for this call.
* sock_var(var, optional) - variable used to store the
RTPProxy socket chosen for this call.
Example 1.17. 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();
};
};
...
1.6.6. rtpproxy_stop_stream2uac([[set_id][, sock_var]]),
rtpproxy_stop_stream2uas([[set_id][, sock_var]])
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.
Meaning of the parameters is as follows:
* set_id(int, optional) - the set used for this call.
* sock_var(var, optional) - variable used to store the
RTPProxy socket chosen for this call.
These functions can be used from REQUEST_ROUTE, ONREPLY_ROUTE.
1.6.7. rtpproxy_start_recording([[set_id][, [sock_var][, [flags][,
[destination][, mediastream]]]]])
This command will send a signal to the RTP-Proxy to record the
RTP stream on the RTP-Proxy.
Meaning of the parameters is as follows:
* set_id(int, optional) - the set used for this call.
* sock_var(var, optional) - variable used to store the
RTPProxy socket chosen for this call.
* flags(string, optional) - a list of flags passed to
RTPProxy for the recording. Currently only s is supported,
and it indicates that RTPProxy should record both audio
legs in a single file. Note that this feature is available
starting with RTPProxy 2.0.
* destination(string, optional) - the destination of the
recording. If it has the udp:IP:port format, RTPProxy sends
the RTP stream to that IP:port remote destination.
Otherwise, destination represents the name of the file in
the recording directory.
* mediastream(int, optional) - this parameter is only used if
the destination is specified, and represents the index of
media stream to record/copy, starting from 1. If this
parameter is missing, OpenSIPS instructs RTPProxy to copy
all the streams.
This function can be used from REQUEST_ROUTE and ONREPLY_ROUTE.
Example 1.18. rtpproxy_start_recording usage
...
rtpproxy_start_recording();
# copy RTP stream to a different listener
rtpproxy_start_recording(,,,"udp:127.0.0.1:60000");
# copy only first RTP stream (audio stream)
rtpproxy_start_recording(,,,"udp:127.0.0.1:60000", 1);
...
1.6.8. rtpproxy_stats(up_pvar, down_var, sent_var, fail_var[,
[set_id][, sock_var]])
This command gathers call RTP statistics from RTP-Proxy.
Meaning of the parameters is as follows:
* up_var (var) - the variable used to return the packets sent
by upstream for this call.
* down_var (var) - the variable used to return the packets
sent by downstream for this call.
* sent_var (var) - the variable used to return the total
number of packets sent for this call.
* up_var (var) - the variable used to return the number of
failed packets for this call.
* set_id(int, optional) - the set used for this call.
* sock_var(var, optional) - variable used to store the
RTPProxy socket chosen for this call.
This function can be used from REQUEST_ROUTE, FAILURE_ROUTE,
ONREPLY_ROUTE, BRANCH_ROUTE and LOCAL_ROUTE.
Example 1.19. rtpproxy_stats usage
...
rtpproxy_stats($var(up),$var(down),$var(sent),$var(fail));
xlog("RTP statistics for $ci: up=$var(up) down=$var(down) sent=$var(sent
) fail=$var(fail)\n");
...
1.6.9. rtpproxy_all_stats(stats_avp[, [set_id][, sock_var]])
This command gathers all RTP statistics available from
RTP-Proxy. All the returned values stored in an AVP that can be
further read by indexing the AVP.
This command is only available starting with RTPProxy 2.1
realease.
Meaning of the parameters is as follows:
* stats_avp (var) - an AVP where the statistics will be
stored. This AVP can be further indexed to get a specific
statistic.
* set_id(int, optional) - the set used for this call.
* sock_var(var, optional) - variable used to store the
RTPProxy socket chosen for this call.
This function can be used from REQUEST_ROUTE, FAILURE_ROUTE,
ONREPLY_ROUTE, BRANCH_ROUTE and LOCAL_ROUTE.
Each statistic is stored at a specific index as it follows:
* ttl - $avp(ret) / $(avp(ret)[0])
* pkts_ia - $(avp(ret)[1])
* pkts_io - $(avp(ret)[2])
* relayed - $(avp(ret)[3])
* dropped - $(avp(ret)[4])
* rtpa_set - $(avp(ret)[5])
* rtpa_rcvd - $(avp(ret)[6])
* rtpa_dups - $(avp(ret)[7])
* rtpa_lost - $(avp(ret)[8])
* rtpa_perrs - $(avp(ret)[9])
Example 1.20. rtpproxy_all_stats usage
...
rtpproxy_all_stats($avp(stats));
xlog("RTP statistics for $ci: dropped=$(avp(stats)[4])\n");
...
1.7. Exported MI Functions
1.7.1. rtpproxy_enable
Enables/Disables a rtp proxy.
Parameters:
* url - the rtp proxy url (exactly as defined in the config
file).
* setid (optional) - the rtpproxy set ID (used for better
indentification of the rtpproxy instance to be enabled, for
example when a rtpproxy is used in multiple sets).
* enable - 1 - enable, 0 - disable. the config file).
NOTE: if a rtpproxy is defined multiple times (in the same or
different set), all its instances will be enables/disabled IF
no set ID provided (as second param).
Example 1.21. rtpproxy_enable usage
...
## disable a RTPProxy by URL only
$ opensips-cli -x mi rtpproxy_enable udp:192.168.2.133:8081 0
## disable a RTPProxy by URL and set ID (3)
$ opensips-cli -x mi rtpproxy_enable udp:192.168.2.133:8081 3 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 1.22. rtpproxy_show usage
...
$ opensips-cli -x mi 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 1.23. rtpproxy_reload usage
...
$ opensips-cli -x mi 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.
1.8.2. E_RTPPROXY_DTMF
This event is raised when a RTPProxy server sends a DTMF
notification to OpenSIPS. In order to catch RFC 2833/4733 DTMF
events, you need to provide the d flag to rtpproxy_offer()/
rtpproxy_answer().
Parameters:
* digit - the digit pressed.
* duration - the duration of the event.
* volume - the volume of the event.
* id - represents the identifier of the call for which that
event was received.
* is_callid - is 0 if the id parameter represents the Dialog
ID, or 1 if it is a callid.
* stream - indicates the stream index of the RTPProxy
session. It is normally 0 if the caller sent the DTMF, or 1
if the callee sent it.
Chapter 2. Frequently Asked Questions
2.1.
What happened 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 https://opensips.org/.
2.3.
Where can I post a question about this module?
First at all check if your question was already answered on one
of our mailing lists:
* User Mailing List -
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
* Developer Mailing List -
http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
E-mails regarding any stable OpenSIPS release should be sent to
<users@lists.opensips.org> and e-mails regarding development
versions should be sent to <devel@lists.opensips.org>.
If you want to keep the mail private, send it to
<users@lists.opensips.org>.
2.4.
How can I report a bug?
Please follow the guidelines provided at:
https://github.com/OpenSIPS/opensips/issues.
Chapter 3. Contributors
3.1. By Commit Statistics
Table 3.1. Top contributors by DevScore^(1), authored
commits^(2) and lines added/removed^(3)
Name DevScore Commits Lines ++ Lines --
1. Razvan Crainea (@razvancrainea) 229 144 5017 2680
2. Maksym Sobolyev (@sobomax) 53 7 5009 256
3. Bogdan-Andrei Iancu (@bogdan-iancu) 27 23 123 116
4. Liviu Chircu (@liviuchircu) 23 18 108 198
5. Vlad Patrascu (@rvlad-patrascu) 23 8 362 699
6. Ovidiu Sas (@ovidiusas) 7 5 31 22
7. Peter Lemenkov (@lemenkov) 6 4 60 11
8. Vlad Paiu (@vladpaiu) 6 4 15 9
9. Ryan Bullock (@rrb3942) 4 2 68 9
10. John Burke (@john08burke) 4 2 8 6
All remaining contributors: robdyck, Dave Sidwell
(@davesidwell), Ezequiel Lovelle (@lovelle), Christophe Sollet
(@csollet), Anca Vamanu, Mikko Lehto, Dan Pascu (@danpascu),
Walter Doekes (@wdoekes), Dusan Klinec (@ph4r05), Julián Moreno
Patiño, Zero King (@l2dy).
(1) DevScore = author_commits + author_lines_added /
(project_lines_added / project_commits) + author_lines_deleted
/ (project_lines_deleted / project_commits)
(2) including any documentation-related commits, excluding
merge commits. Regarding imported patches/code, we do our best
to count the work on behalf of the proper owner, as per the
"fix_authors" and "mod_renames" arrays in
opensips/doc/build-contrib.sh. If you identify any
patches/commits which do not get properly attributed to you,
please submit a pull request which extends "fix_authors" and/or
"mod_renames".
(3) ignoring whitespace edits, renamed files and auto-generated
files
3.2. By Commit Activity
Table 3.2. Most recently active contributors^(1) to this module
Name Commit Activity
1. Razvan Crainea (@razvancrainea) Mar 2011 - Nov 2021
2. Liviu Chircu (@liviuchircu) Jul 2012 - Sep 2021
3. John Burke (@john08burke) Apr 2021 - Apr 2021
4. Maksym Sobolyev (@sobomax) Mar 2011 - Jan 2021
5. Peter Lemenkov (@lemenkov) Dec 2011 - Jul 2020
6. Ovidiu Sas (@ovidiusas) Mar 2011 - Jun 2020
7. Bogdan-Andrei Iancu (@bogdan-iancu) Mar 2011 - Apr 2020
8. robdyck Apr 2020 - Apr 2020
9. Zero King (@l2dy) Mar 2020 - Mar 2020
10. Vlad Patrascu (@rvlad-patrascu) May 2017 - Nov 2019
All remaining contributors: Dan Pascu (@danpascu), Ryan Bullock
(@rrb3942), Julián Moreno Patiño, Vlad Paiu (@vladpaiu), Dusan
Klinec (@ph4r05), Dave Sidwell (@davesidwell), Ezequiel Lovelle
(@lovelle), Mikko Lehto, Walter Doekes (@wdoekes), Christophe
Sollet (@csollet), Anca Vamanu.
(1) including any documentation-related commits, excluding
merge commits
Chapter 4. Documentation
4.1. Contributors
Last edited by: Razvan Crainea (@razvancrainea), Liviu Chircu
(@liviuchircu), Maksym Sobolyev (@sobomax), Zero King (@l2dy),
Vlad Patrascu (@rvlad-patrascu), Peter Lemenkov (@lemenkov),
Julián Moreno Patiño, Bogdan-Andrei Iancu (@bogdan-iancu),
Mikko Lehto, Ryan Bullock (@rrb3942), Ovidiu Sas (@ovidiusas).
Documentation Copyrights:
Copyright © 2005 Voice Sistem SRL
Copyright © 2003-2008 Sippy Software, Inc.
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