Copyright © 2003-2008 Sippy Software, Inc.
Copyright © 2005 voice-system.ro
| Revision History | |
|---|---|
| Revision $Revision: 5901 $ | $Date: 2009-07-21 09:45:05 +0200 (Tue, 21 Jul 2009) $ |
Table of Contents
natping_interval (integer)ping_nated_only (integer)natping_processes (integer)natping_socket (string)received_avp (str)rtpproxy_sock (string)rtpproxy_disable_tout (integer)rtpproxy_timeout (string)rtpproxy_tout (integer)rtpproxy_retr (integer)force_socket (string)sipping_bflag (integer)sipping_from (string)sipping_method (string)nortpproxy_str (string)db_url (string)db_table (string)rtpp_socket_col (string)set_id_col (string)rtpp_notify_socket (string)fix_nated_contact([uri_params])
fix_nated_sdp(flags [, ip_address])
set_rtp_proxy_set()
rtpproxy_offer([flags [, ip_address]])
rtpproxy_answer([flags [, ip_address]])
unforce_rtp_proxy()
rtpproxy_stream2uac(prompt_name, count),
rtpproxy_stream2uas(prompt_name, count)
rtpproxy_stop_stream2uac(),
rtpproxy_stop_stream2uas()
add_rcv_param([flag]),
fix_nated_register()
nat_uac_test(flags)
start_recording()
List of Examples
natping_interval parameterping_nated_only parameternatping_processes parameternatping_socket parameterreceived_avp parameterrtpproxy_sock parameterrtpproxy_disable_tout parameterrtpproxy_timeout parameter to 200msrtpproxy_retr parameterforce_socket parametersipping_bflag parametersipping_from parametersipping_method parameternortpproxy_str parameterdb_url parameterdb_table parameterrtpp_socket_col parameterset_id parameterrtpp_notify_socket parameterfix_nated_contact usagefix_nated_sdp usagefix_nated_contact usagertpproxy_offer usagertpproxy_answer usageunforce_rtp_proxy usagertpproxy_stream2xxx usageadd_rcv_paramer usagefix_nated_register usagestart_recording usagenh_enable_ping usagenh_enable_rtpp usagenh_show_rtpp usagenh_reload_rtpp usageThis 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).
Works with multipart messages that contain an SDP part, but not with multi-layered multipart messages.
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;
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. 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.
The selection of the set is done from script prior using unforce_rtp_proxy(), rtpproxy_offer() or rtpproxy_answer() functions - see the set_rtp_proxy_set() function.
For backward compatibility reasons, a set with no id take by default the id 0. Also if no set is explicitly set before unforce_rtp_proxy(), rtpproxy_offer() or rtpproxy_answer() the 0 id set will be used.
IMPORTANT: if you use multiple sets, take care and use the same set for both rtpproxy_offer()/rtpproxy_answer() and unforce_rtpproxy()!!
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 nathelper 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.
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_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: http://opensips.org/pub/rtpproxy/
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.
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.
Enabling the NAT pinging functionality will force the module to bind itself to USRLOC module.
Default value is 0.
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.
How many timer processes should be created by the module for the exclusive task of sending the NAT pings.
Default value is 1.
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")
...
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.
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).
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 with specific weight
modparam("nathelper", "rtpproxy_sock", "udp:localhost:12221=2")
# 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")
...
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)
...
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")
...
How many times nathelper should retry to send and receive after timeout was generated.
Default value is “5”.
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")
...
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).
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")
...
The parameter sets the SIP method to be used in generating the SIP requests for NAT ping purposes.
Default value is “OPTIONS”.
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.
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")
...
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")
...
The name of the database table containing definitions of socket(s) used to connect to (a set) RTPProxy.
Default value is “nh_sockets”.
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")
...
The name set id column in the database table.
Default value is “set_id”.
Rewrites the URI Contact HF to contain request's source address:port. If a list of URI parameter is provided, it will be added to the modified contact;
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE.
Example 1.20. fix_nated_contact usage
...
if (search("User-Agent: Cisco ATA.*") {
fix_nated_contact(";ata=cisco");
} else {
fix_nated_contact();
}
...
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 1.21. fix_nated_sdp usage
...
if (search("User-Agent: Cisco ATA.*") {fix_nated_sdp("3");};
...
Sets the Id of the rtpproxy set to be used for the next unforce_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.
Rewrites SDP body to ensure that media is passed through an RTP proxy. To be invoked on INVITE for the cases the SDPs are in INVITE and 200 OK and on 200 OK when SDPs are in 200 OK and ACK.
Meaning of the parameters is as follows:
flags - flags to turn on some features.
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.
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 1.23. 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();
...
}
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.
Tears down the RTPProxy session for the current call.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE.
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 - 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 ($rb=~ "0\.0\.0\.0") {
rtpproxy_stream2uas("/var/rtpproxy/prompts/music_on_hold", "-1");
} else {
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.
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
...
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.
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.
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 $ ...
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 ...
Displays all the rtp proxies and their information: set and status (disabled or not, weight and recheck_ticks).
No parameter.
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.
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:
E-mails regarding any stable OpenSIPS release should be sent to
If you want to keep the mail private, send it to
| |
2.4. | How can I report a bug? |
Please follow the guidelines provided at: https://github.com/OpenSIPS/opensips/issues. |