nathelper Module
Maxim Sobolev
Sippy Software, Inc.
Edited by
Maxim Sobolev
Edited by
Bogdan-Andrei Iancu
Copyright © 2003-2008 Sippy Software, Inc.
Copyright © 2005 voice-system.ro
Revision History
Revision $Revision: 6684 $ $Date: 2009-07-21 10:45:05 +0300
(Tue, 21 Jul 2009) $
__________________________________________________________
Table of Contents
1. Admin Guide
1.1. Overview
1.2. NAT pinging types
1.3. Multiple RTPProxy usage
1.4. Dependencies
1.4.1. OpenSIPS Modules
1.4.2. External Libraries or Applications
1.5. Exported Parameters
1.5.1. natping_interval (integer)
1.5.2. ping_nated_only (integer)
1.5.3. natping_processes (integer)
1.5.4. natping_socket (string)
1.5.5. received_avp (str)
1.5.6. rtpproxy_sock (string)
1.5.7. rtpproxy_disable_tout (integer)
1.5.8. rtpproxy_timeout (string)
1.5.9. rtpproxy_tout (integer)
1.5.10. rtpproxy_retr (integer)
1.5.11. force_socket (string)
1.5.12. sipping_bflag (integer)
1.5.13. sipping_from (string)
1.5.14. sipping_method (string)
1.5.15. nortpproxy_str (string)
1.5.16. db_url (string)
1.5.17. db_table (string)
1.5.18. rtpp_socket_col (string)
1.5.19. set_id_col (string)
1.6. Exported Functions
1.6.1. fix_nated_contact()
1.6.2. fix_nated_sdp(flags [, ip_address])
1.6.3. set_rtp_proxy_set()
1.6.4. force_rtp_proxy([flags [, ip_address]])
1.6.5. rtpproxy_offer([flags [, ip_address]])
1.6.6. rtpproxy_answer([flags [, ip_address]])
1.6.7. unforce_rtp_proxy()
1.6.8. rtpproxy_stream2uac(prompt_name, count),
rtpproxy_stream2uas(prompt_name, count)
1.6.9. rtpproxy_stop_stream2uac(),
rtpproxy_stop_stream2uas()
1.6.10. add_rcv_param([flag]),
1.6.11. fix_nated_register()
1.6.12. nat_uac_test(flags)
1.6.13. start_recording()
1.7. MI Commands
1.7.1. nh_enable_ping
1.7.2. nh_enable_rtpp
1.7.3. nh_show_rtpp
1.7.4. nh_reload_rtpp
2. Frequently Asked Questions
List of Examples
1.1. Set natping_interval parameter
1.2. Set ping_nated_only parameter
1.3. Set natping_processes parameter
1.4. Set natping_socket parameter
1.5. Set received_avp parameter
1.6. Set rtpproxy_sock parameter
1.7. Set rtpproxy_disable_tout parameter
1.8. Set rtpproxy_timeout parameter to 200ms
1.9. Set rtpproxy_retr parameter
1.10. Set force_socket parameter
1.11. Set sipping_bflag parameter
1.12. Set sipping_from parameter
1.13. Set sipping_method parameter
1.14. Set nortpproxy_str parameter
1.15. Set db_url parameter
1.16. Set db_table parameter
1.17. Set rtpp_socket_col parameter
1.18. Set set_id parameter
1.19. fix_nated_contact usage
1.20. fix_nated_sdp usage
1.21. fix_nated_contact usage
1.22. force_rtp_proxy usage
1.23. rtpproxy_offer usage
1.24.
1.25. unforce_rtp_proxy usage
1.26. rtpproxy_stream2xxx usage
1.27. add_rcv_paramer usage
1.28. fix_nated_register usage
1.29. start_recording usage
1.30. nh_enable_ping usage
1.31. nh_enable_rtpp usage
1.32. nh_show_rtpp usage
1.33. nh_reload_rtpp usage
Chapter 1. Admin Guide
1.1. Overview
This is a module to help with NAT traversal. In particular, it
helps symmetric UAs that don't advertise they are symmetric and
are not able to determine their public address.
fix_nated_contact rewrites Contact header field with request's
source address:port pair. fix_nated_sdp adds the active
direction indication to SDP (flag 0x01) and updates source IP
address too (flag 0x02).
Known devices that get along over NATs with nathelper are ATAs
(as clients) and Cisco Gateways (since 12.2(T)) as servers. See
http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products
_feature_guide09186a0080110bf9.html">
Works with multipart messages that contain an SDP part, but not
with multi-layered multipart messages.
1.2. NAT pinging types
Currently, the nathelper module supports two types of NAT
pings:
* UDP package - 4 bytes (zero filled) UDP packages are sent
to the contact address.
+ Advantages: low bandwitdh traffic, easy to generate by
OpenSIPS;
+ Disadvantages: unidirectional traffic through NAT
(inbound - from outside to inside); As many NATs do
update the bind timeout only on outbound traffic, the
bind may expire and closed.
* SIP request - a stateless SIP request is sent to the
contact address.
+ Advantages: bidirectional traffic through NAT, since
each PING request from OpenSIPS (inbound traffic) will
force the SIP client to generate a SIP reply (outbound
traffic) - the NAT bind will be surely kept open.
+ Disadvantages: higher bandwitdh traffic, more
expensive (as time) to generate by OpenSIPS;
1.3. Multiple RTPProxy usage
Currently, the nathelper 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.
The selection of the set is done from script prior using
[un]force_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
[un]force_rtp_proxy(), rtpproxy_offer() or rtpproxy_answer()
the 0 id set will be used.
IMPORTANT: if you use nultiple sets, take care and use the same
set for both force_ and unforce_rtpproxy()!!
1.4. Dependencies
1.4.1. OpenSIPS Modules
The following modules must be loaded before this module:
* usrloc module - only if the NATed contacts are to be
pinged.
* a database module - only if you want to load use a database
table from where to load the rtp proxies sets.
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. natping_interval (integer)
Period of time in seconds between sending the NAT pings to all
currently registered UAs to keep their NAT bindings alive.
Value of 0 disables this functionality.
Note
Enabling the NAT pinging functionality will force the module to
bind itself to USRLOC module.
Default value is 0.
Example 1.1. Set natping_interval parameter
...
modparam("nathelper", "natping_interval", 10)
...
1.5.2. ping_nated_only (integer)
If this variable is set then only contacts that have
"behind_NAT" flag in user location database set will get ping.
Default value is 0.
Example 1.2. Set ping_nated_only parameter
...
modparam("nathelper", "ping_nated_only", 1)
...
1.5.3. natping_processes (integer)
How many timer processes should be created by the module for
the exclusive task of sending the NAT pings.
Default value is 1.
Example 1.3. Set natping_processes parameter
...
modparam("nathelper", "natping_processes", 3)
...
1.5.4. natping_socket (string)
Spoof the natping's source-ip to this address. Works only for
IPv4.
Default value is NULL.
Example 1.4. Set natping_socket parameter
...
modparam("nathelper", "natping_socket", "192.168.1.1:5006")
...
1.5.5. received_avp (str)
The name of the Attribute-Value-Pair (AVP) used to store the
URI containing the received IP, port, and protocol. The URI is
created by fix_nated_register function of nathelper module and
the attribute is then used by the registrar to store the
received parameters. Do not forget to change the value of
corresponding parameter in registrar module if you change the
value of this parameter.
Note
You must set this parameter if you use "fix_nated_register". In
such case you must set the parameter with same name of
"registrar" module to same value.
Default value is "NULL" (disabled).
Example 1.5. Set received_avp parameter
...
modparam("nathelper", "received_avp", "$avp(i:42)")
...
1.5.6. 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 1.6. Set rtpproxy_sock parameter
...
# single rtproxy
modparam("nathelper", "rtpproxy_sock", "udp:localhost:12221")
# multiple rtproxies for LB
modparam("nathelper", "rtpproxy_sock",
"udp:localhost:12221 udp:localhost:12222")
# multiple sets of multiple rtproxies
modparam("nathelper", "rtpproxy_sock",
"1 == udp:localhost:12221 udp:localhost:12222")
modparam("nathelper", "rtpproxy_sock",
"2 == udp:localhost:12225")
...
1.5.7. rtpproxy_disable_tout (integer)
Once RTPProxy was found unreachable and marked as disable,
nathelper will not attempt to establish communication to
RTPProxy for rtpproxy_disable_tout seconds.
Default value is "60".
Example 1.7. Set rtpproxy_disable_tout parameter
...
modparam("nathelper", "rtpproxy_disable_tout", 20)
...
1.5.8. rtpproxy_timeout (string)
Timeout value in waiting for reply from RTPProxy.
Default value is "1".
Example 1.8. Set rtpproxy_timeout parameter to 200ms
...
modparam("nathelper", "rtpproxy_timeout", "0.2")
...
1.5.9. rtpproxy_tout (integer)
Obsolete. see rtpproxy_timeout.
1.5.10. rtpproxy_retr (integer)
How many times nathelper should retry to send and receive after
timeout was generated.
Default value is "5".
Example 1.9. Set rtpproxy_retr parameter
...
modparam("nathelper", "rtpproxy_retr", 2)
...
1.5.11. force_socket (string)
Socket to be forced in communicating to RTPProxy. It makes
sense only for UDP communication. If no one specified, the OS
will choose.
Default value is "NULL".
Example 1.10. Set force_socket parameter
...
modparam("nathelper", "force_socket", "localhost:33333")
...
1.5.12. sipping_bflag (integer)
What branch flag should be used by the module to identify NATed
contacts for which it should perform NAT ping via a SIP request
instead if dummy UDP package.
Default value is -1 (disabled).
Example 1.11. Set sipping_bflag parameter
...
modparam("nathelper", "sipping_bflag", 7)
...
1.5.13. sipping_from (string)
The parameter sets the SIP URI to be used in generating the SIP
requests for NAT ping purposes. To enable the SIP request
pinging feature, you have to set this parameter. The SIP
request pinging will be used only for requests marked so.
Default value is "NULL".
Example 1.12. Set sipping_from parameter
...
modparam("nathelper", "sipping_from", "sip:pinger@siphub.net")
...
1.5.14. sipping_method (string)
The parameter sets the SIP method to be used in generating the
SIP requests for NAT ping purposes.
Default value is "OPTIONS".
Example 1.13. Set sipping_method parameter
...
modparam("nathelper", "sipping_method", "INFO")
...
1.5.15. nortpproxy_str (string)
The parameter sets the SDP attribute used by nathelper 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 1.14. Set nortpproxy_str parameter
...
modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n")
...
1.5.16. 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 nh_reload_rtpp is issued(the definitions will be
replaced with the ones from the database table).
Default value is "NULL".
Example 1.15. Set db_url parameter
...
modparam("nathelper", "db_url",
"mysql://opensips:opensipsrw@192.168.2.132/opensips")
...
1.5.17. db_table (string)
The name of the database table containing definitions of
socket(s) used to connect to (a set) RTPProxy.
Default value is "nh_sockets".
Example 1.16. Set db_table parameter
...
modparam("nathelper", "db_table", "nh_rtpp")
...
1.5.18. rtpp_socket_col (string)
The name rtpp socket column in the database table.
Default value is "rtpproxy_sock".
Example 1.17. Set rtpp_socket_col parameter
...
modparam("nathelper", "rtpp_socket_col", "rtpp_socket")
...
1.5.19. set_id_col (string)
The name set id column in the database table.
Default value is "set_id".
Example 1.18. Set set_id parameter
...
modparam("nathelper", "set_id_col", "rtpp_set_id")
...
1.6. Exported Functions
1.6.1. fix_nated_contact()
Rewrites Contact HF to contain request's source address:port.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
BRANCH_ROUTE.
Example 1.19. fix_nated_contact usage
...
if (search("User-Agent: Cisco ATA.*") {fix_nated_contact();};
...
1.6.2. fix_nated_sdp(flags [, ip_address])
Alters the SDP information in orer to facilitate NAT traversal.
What changes to be performed may be controled via the "flags"
paramter.
Meaning of the parameters is as follows:
* flags - the value may be a bitwise OR of the following
flags:
+ 0x01 - adds "a=direction:active" SDP line;
+ 0x02 - rewrite media IP address (c=) with source
address of the message or the provided IP address (the
provide IP address take precedence over the source
address).
+ 0x04 - adds "a=nortpproxy:yes" SDP line;
+ 0x08 - rewrite IP from origin description (o=) with
source address of the message or the provided IP
address (the provide IP address take precedence over
the source address).
* ip_address - IP to be used for rewritting SDP. If not
specified, the received signalling IP will be used. The
parameter allows pseudo-variables usage. NOTE: For the IP
to be used, you need to use 0x02 or 0x08 flags, otherwise
it will have no effect.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
FAILURE_ROUTE, BRANCH_ROUTE.
Example 1.20. fix_nated_sdp usage
...
if (search("User-Agent: Cisco ATA.*") {fix_nated_sdp("3");};
...
1.6.3. set_rtp_proxy_set()
Sets the Id of the rtpproxy set to be used for the next
[un]force_rtp_proxy(), rtpproxy_offer() or rtpproxy_answer()
command.
Paramter can also be a pseudo-variable that contain (as string
or integer) the ID of the rtpproxy set to be used.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
BRANCH_ROUTE.
Example 1.21. fix_nated_contact usage
...
set_rtp_proxy_set("2");
force_rtp_proxy();
...
1.6.4. force_rtp_proxy([flags [, ip_address]])
Rewrites SDP body to ensure that media is passed through an RTP
proxy. It can have optional parameters to force additional
features. If ip_address is provided, it will be used to replace
the one in SDP.
The function is considered depreciated and provided for the
compatibility purposes. Use rtpproxy_offer() or
rtpproxy_answer() instead.
Meaning of the parameters is as follows:
* flags - flags to turn on some features.
+ a - flags that UA from which message is received
doesn't support symmetric RTP.
+ l - force "lookup", that is, only rewrite SDP when
corresponding session is already exists in the RTP
proxy. By default is on when the session is to be
completed (reply in non-swap or ACK in swap mode).
+ i - flags that message is received from UA in the LAN
(internal network). Only makes sense when RTP proxy is
running in the bridge mode.
+ e - flags that message is received from UA in the WAN
(external network). Only makes sense when RTP proxy is
running in the bridge mode.
+ f - instructs nathelper to ignore marks inserted by
another nathelper 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, nathelper 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 - flags to swap creation with confirmation between
requests and replies. By default, a request creates
the RTP session and a reply confirms it. If swapped, a
reply will create the RTP session and a request will
confirm it. The flag is considered depreciated and
provided for the compatibility purposes. Use
rtpproxy_offer() or rtpproxy_answer() instead.
+ 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 REQUEST_ROUTE, ONREPLY_ROUTE,
FAILURE_ROUTE, BRANCH_ROUTE.
Example 1.22. force_rtp_proxy usage
...
if (search("User-Agent: Cisco ATA.*") {force_rtp_proxy();};
if (src_ip=1.2.3.4) {force_rtp_proxy("i");};
if (search("User-Agent: Cisco ATA.*") {force_rtp_proxy("","1.2.3.4");};
...
1.6.5. rtpproxy_offer([flags [, ip_address]])
Rewrites SDP body to ensure that media is passed through an RTP
proxy. Equivalent of force_rtp_proxy() function to be invoked
on INVITE for the cases the SDPs are in INVITE and 200 OK and
on 200 OK when SDPs are in 200 OK and ACK.
See force_rtp_proxy() function description above for the
meaning of the parameters.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
FAILURE_ROUTE, BRANCH_ROUTE.
Example 1.23. rtpproxy_offer usage
route {
...
if (is_method("INVITE")) {
if (has_sdp()) {
if (rtpproxy_offer())
t_on_reply("1");
} else {
t_on_reply("2");
}
}
if (is_method("ACK") && has_sdp())
rtpproxy_answer();
...
}
onreply_route[1]
{
...
if (has_sdp())
rtpproxy_answer();
...
}
onreply_route[2]
{
...
if (has_sdp())
rtpproxy_offer();
...
}
1.6.6. rtpproxy_answer([flags [, ip_address]])
Rewrites SDP body to ensure that media is passed through an RTP
proxy. Equivalent of force_rtp_proxy() function 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 force_rtp_proxy() function description above for the
meaning of the parameters.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
FAILURE_ROUTE, BRANCH_ROUTE.
Example 1.24.
See rtpproxy_offer() function example above for example.
1.6.7. unforce_rtp_proxy()
Tears down the RTPProxy session for the current call.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
FAILURE_ROUTE, BRANCH_ROUTE.
Example 1.25. unforce_rtp_proxy usage
...
unforce_rtp_proxy();
...
1.6.8. rtpproxy_stream2uac(prompt_name, count),
rtpproxy_stream2uas(prompt_name, count)
Instruct the RTPproxy to stream prompt/announcement pre-encoded
with the makeann command from the RTPproxy distribution. The
uac/uas suffix selects who will hear the announcement
relatively to the current transaction - UAC or UAS. For example
invoking the rtpproxy_stream2uac in the request processing
block on ACK transaction will play the prompt to the UA that
has generated original INVITE and ACK while
rtpproxy_stop_stream2uas on 183 in reply processing block will
play the prompt to the UA that has generated 183.
Apart from generating announcements, another possible
application of this function is implementing music on hold
(MOH) functionality. When count is -1, the streaming will be in
loop indefinitely until the appropriate
rtpproxy_stop_stream2xxx is issued.
In order to work correctly, functions require that the session
in the RTPproxy already exists. Also those functions don't
alted SDP, so that they are not substitute for calling
rtpproxy_offer, rtpproxy_answer or force_rtp_proxy.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE.
Meaning of the parameters is as follows:
* prompt_name - name of the prompt to stream. Should be
either absolute pathname or pathname relative to the
directory where RTPproxy runs.
* count - number of times the prompt should be repeated. The
value of -1 means that it will be streaming in loop
indefinitely, until appropriate rtpproxy_stop_stream2xxx is
issued.
Example 1.26. 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();
};
};
...
1.6.9. rtpproxy_stop_stream2uac(), rtpproxy_stop_stream2uas()
Stop streaming of announcement/prompt/MOH started previously by
the respective rtpproxy_stream2xxx. The uac/uas suffix selects
whose announcement relatively to tha current transaction should
be stopped - UAC or UAS.
These functions can be used from REQUEST_ROUTE, ONREPLY_ROUTE.
1.6.10. add_rcv_param([flag]),
Add received parameter to Contact header fields or Contact URI.
The parameter will contain URI created from the source IP,
port, and protocol of the packet containing the SIP message.
The parameter can be then processed by another registrar, this
is useful, for example, when replicating register messages
using t_replicate function to another registrar.
Meaning of the parameters is as follows:
* flag - flags to indicate if the parameter should be added
to Contact URI or Contact header. If the flag is non-zero,
the parameter will be added to the Contact URI. If not used
or equal to zero, the parameter will go to the Contact
header.
This function can be used from REQUEST_ROUTE.
Example 1.27. add_rcv_paramer usage
...
add_rcv_param(); # add the parameter to the Contact header
....
add_rcv_param("1"); # add the paramter to the Contact URI
...
1.6.11. fix_nated_register()
The function creates a URI consisting of the source IP, port,
and protocol and stores the URI in an Attribute-Value-Pair. The
URI will be appended as "received" parameter to Contact in 200
OK and registrar will store it in the user location database.
This function can be used from REQUEST_ROUTE.
Example 1.28. fix_nated_register usage
...
fix_nated_register();
...
1.6.12. nat_uac_test(flags)
Tries to guess if client's request originated behind a nat. The
parameter determines what heuristics is used.
Meaning of the flags is as follows:
* 1 - Contact header field is searched for occurrence of
RFC1918 addresses.
* 2 - the "received" test is used: address in Via is compared
against source IP address of signaling
* 4 - Top Most VIA is searched for occurrence of RFC1918
addresses
* 8 - SDP is searched for occurrence of RFC1918 addresses
* 16 - test if the source port is different from the port in
Via
* 32 - address in Contact is compared against source IP
address of signaling
All flags can be bitwise combined, the test returns true if any
of the tests identified a NAT.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
FAILURE_ROUTE, BRANCH_ROUTE.
1.6.13. start_recording()
This command will send a signal to the RTP-Proxy to record the
RTP stream on the RTP-Proxy.
This function can be used from REQUEST_ROUTE and ONREPLY_ROUTE.
Example 1.29. start_recording usage
...
start_recording();
...
1.7. MI Commands
1.7.1. nh_enable_ping
Enables natping if parameter value greater than 0. Disables
natping if parameter value is 0. With no parameter, it returns
the current natping status.
The function may takean optional parameter (for set operations)
- a number in decimal format.
Example 1.30. nh_enable_ping usage
...
$ opensipsctl fifo nh_enable_ping
Status:: 1
$
$ opensipsctl fifo nh_enable_ping 0
$
$ opensipsctl fifo nh_enable_ping
Status:: 0
$
...
1.7.2. 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 its instances will be enables/disabled.
Example 1.31. nh_enable_rtpp usage
...
$ opensipsctl fifo nh_enable_rtpp udp:192.168.2.133:8081 0
...
1.7.3. nh_show_rtpp
Displays all the rtp proxies and their information: set and
status (disabled or not, weight and recheck_ticks).
No parameter.
Example 1.32. nh_show_rtpp usage
...
$ opensipsctl fifo nh_show_rtpp
...
1.7.4. nh_reload_rtpp
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 1.33. nh_reload_rtpp usage
...
$ opensipsctl fifo nh_reload_rtpp
...
Chapter 2. Frequently Asked Questions
2.1.
What happend with "rtpproxy_disable" parameter?
It was removed as it became obsolete - now "rtpproxy_sock" can
take empty value to disable the rtpproxy functionality.
2.2.
Where can I find more about OpenSIPS?
Take a look at http://www.opensips.org/.
2.3.
Where can I post a question about this module?
First at all check if your question was already answered on one
of our mailing lists:
* User Mailing List -
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
* Developer Mailing List -
http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
E-mails regarding any stable OpenSIPS release should be sent to
<users@lists.opensips.org> and e-mails regarding development
versions should be sent to <devel@lists.opensips.org>.
If you want to keep the mail private, send it to
<users@lists.opensips.org>.
2.4.
How can I report a bug?
Please follow the guidelines provided at:
http://sourceforge.net/tracker/?group_id=232389.
