Copyright © 2003-2008 Sippy Software, Inc.
Copyright © 2005 Voice Sistem SRL
Copyright © 2009 TuTPro Inc.
Copyright © 2010 VoIPEmbedded Inc.
Table of Contents
force_socket (string)
natping_interval (integer)
ping_nated_only (integer)
natping_processes (integer)
natping_socket (string)
received_avp (str)
sipping_bflag (integer)
sipping_from (string)
sipping_method (string)
natping_disable_bflag (integer)
nortpproxy_str (string)
keepalive_timeout (int)
udpping_from_path (int)
append_sdp_oldmediaip (int)
filter_server_id (int)
nat_addr_mode (int)
fix_nated_contact()
fix_nated_sdp(flags [, ip_address])
add_rcv_param([flag]),
fix_nated_register()
nat_uac_test(flags)
is_rfc1918(ip_address)
add_contact_alias([ip_addr, port, proto])
handle_ruri_alias()
set_contact_alias([trim])
set_alias_to_pv(target_avp)
List of Examples
force_socket parameternatping_interval parameterping_nated_only parameternatping_processes parameternatping_socket parameterreceived_avp parametersipping_bflag parametersipping_from parametersipping_method parameternatping_disable_bflag parameternortpproxy_str parameterkeepalive_timeout parameterudpping_from_path parameterappend_sdp_oldmediaip parameterfilter_server_id parameternat_addr_mode parameterfix_nated_contact usagefix_nated_sdp usageadd_rcv_paramer usagefix_nated_register usagenat_uac_test usageis_rfc1918 usageadd_contact_alias usagehandle_ruri_alias usageset_contact_alias usageset_alias_to_pv usagenathelper.enable_ping usageTable of Contents
force_socket (string)
natping_interval (integer)
ping_nated_only (integer)
natping_processes (integer)
natping_socket (string)
received_avp (str)
sipping_bflag (integer)
sipping_from (string)
sipping_method (string)
natping_disable_bflag (integer)
nortpproxy_str (string)
keepalive_timeout (int)
udpping_from_path (int)
append_sdp_oldmediaip (int)
filter_server_id (int)
nat_addr_mode (int)
fix_nated_contact()
fix_nated_sdp(flags [, ip_address])
add_rcv_param([flag]),
fix_nated_register()
nat_uac_test(flags)
is_rfc1918(ip_address)
add_contact_alias([ip_addr, port, proto])
handle_ruri_alias()
set_contact_alias([trim])
set_alias_to_pv(target_avp)
This is a module to help with NAT traversal and reuse of TCP connections. In particular, it helps symmetric UAs that don't advertise they are symmetric and are not able to determine their public address.
The function fix_nated_contact() rewrites the “Contact”
header field with request's source address:port pair. The function
fix_nated_sdp() adds the active direction indication
to SDP (flag 0x01) and updates the source IP address too (flag 0x02). The function
fix_nated_register() exports exports the request's source
address:port into an AVP to be used during save() and should
be used for “REGISTER” requests.
Note: fix_nated_contact changes the “Contact”
header, thus it breaks the RFC. Although usually this is not an issue, it may
cause problems with strict SIP clients. An alternative is to use
add_contact_alias() that together with
the handle_ruri_alias() is standards conforming and also
supports reuse of TCP/TLS connections.
Currently, the nathelper module supports two types of NAT pings:
UDP packet - 4 bytes (zero filled) UDP packets are sent to the contact address.
Advantages: low bandwidth traffic, easy to generate by Kamailio;
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 UDP contact address.
Advantages: bidirectional traffic through NAT, since each PING request from Kamailio (inbound traffic) will force the SIP client to generate a SIP reply (outbound traffic) - the NAT bind will be surely kept open.
Disadvantages: higher bandwidth traffic, more expensive (as time) to generate by Kamailio;
The following modules must be loaded before this module:
usrloc module - only if the NATed contacts are to be pinged.
Socket to be used when sending NAT pings for UDP communication. If no one specified, the OS will choose a socket.
Default value is “NULL”.
Example 1.1. Set force_socket parameter
...
modparam("nathelper", "force_socket", "127.0.0.1:5060")
...
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 parameter is set to 1 then only contacts that have the behind NAT “nat_bflag” flag in user location database set will get ping.
If it is 0 and sipping_bflag is not set, then the 4-bytes UDP ping is sent to all contacts. If it is 0 and sipping_bflag parameter is set, then SIP-request-based pinging is sent to all contacts.
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 1.5. 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 in the “registrar”
module to same value.
Default value is "NULL" (disabled).
Which branch flag should be used by the module to identify NATed contacts for which it should perform NAT ping via a SIP request instead of dummy UDP packet.
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 1.8. 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”.
What branch flag should be used by the module to disable NAT pings on a per-registration basis. If the given flag is set for a particular registration, then no NAT pings will be sent at all, regardless of any other conditions.
Default value is -1 (disabled).
Example 1.10. Set natping_disable_bflag parameter
...
modparam("nathelper", "natping_disable_bflag", 8)
...
The parameter sets the SDP attribute used by nathelper to mark the packet SDP that information 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 1.11. Set nortpproxy_str parameter
...
modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n")
...
The parameter sets the interval in seconds after which a natted contact is removed from location table if it does not reply to SIP keepalives (usually OPTIONS ping requests).
The features is available only for UDP contacts that are stored in memory (not working for db only mode for usrloc module).
Keepalives are sent stateless, not using TM module. The value of this parameter has to be few times higher than natping_interval.
Default value is “0” (feature disabled).
Example 1.12. Set keepalive_timeout parameter
...
modparam("nathelper", "keepalive_timeout", 120)
...
Enable sending UDP pings (keepalives) using raw socket from Path address.
Default value is “0” (feature disabled).
The parameter controls if oldmediaip field should be appended to the SDP.
Default value is “1” (feature enabled).
Example 1.14. Set append_sdp_oldmediaip parameter
...
modparam("nathelper", "append_sdp_oldmediaip", 1)
...
Filter contacts by “server_id” core parameter. Use this parameter to limit pinging. When set to “1”, only proxy instances which send packets are those where core server_id matches server_id saved in usrloc. Default value is “0” (disabled).
If set to 0, only default private net addresses are checked by nat_uac_test(). If set to 1, other reserved net addresses are checked by nat_uac_test() as well.
Default private net addresses are:
10.0.0.0/8
172.16.0.0/12
192.168.0.0/16
100.64.0.0/10 - RFC6598 - Carrier Grade NAT
192.0.0.0/29 - RFC7335 - IPv4 Service Continuity Prefix
Reserved net addresses are:
192.0.0.0/24 - RFC7335 - IETF Protocol Assignments
Default value is “1”.
Rewrites the “Contact” header to contain the request's source address:port.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE.
Example 1.17. fix_nated_contact usage
...
if (search("User-Agent: Cisco ATA.*") {fix_nated_contact();};
...
Alters the SDP information in order to facilitate NAT traversal. What changes to be performed may be controlled via the “flags” parameter. Return value is -1 if error occurred, 1 if ip's were replaced, 2 if no ip's were replaced.
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. (a=rtcp) param will be rewritten if exists. (the provided 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. (a=rtcp) param will be rewritten if exists. (the provided IP address take precedence over the source address).
ip_address - IP to be used for rewriting 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. Must be IPv4 address family.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE.
Example 1.18. fix_nated_sdp usage
...
if (search("User-Agent: Cisco ATA.*") {fix_nated_sdp("3");};
...
Add a received parameter to the “Contact” header fields (available for all transports) or to the Contact URI (available only for UDP traffic).
The parameter will contain the URI created from the
source IP, port, and protocol (if different than UDP) 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 1.19. add_rcv_paramer usage
...
add_rcv_param(); # add the parameter to the Contact header
....
add_rcv_param("1"); # add the parameter 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 received column in the location table.
Note: You have to set the “received_avp” parameter of the nathelper module and the registrar module (both module parameters must have the same value) to use this function.
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 - The “Contact” header field is searched for occurrence of RFC1918 or RFC6598 addresses.
2 - the "received" test is used: address in the “Via” header is compared against source IP address of signaling. If the “Via” header contains no port, it uses the default SIP port 5060
4 - The Top Most “Via” is searched for occurrence of RFC1918 or RFC6598 addresses
8 - The SDP is searched for occurrence of RFC1918 or RFC6598 addresses
16 - Test if the source port is different from the port in the “Via” header. If the “Via” header contains no port, it uses the default SIP port 5060
32 - Test if the source IP address of signaling is a RFC1918 or RFC6598 address
64 - Test if the source connection of signaling is a WebSocket
128 - Test if the “Contact” header URI port differs from the source port of the request (Warning: this is might be legal or even intended combination in non NATted scenarios)
256 - Test if the SDP connection address is different from source IP address. It will work also with multiple connection address lines.
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.
Determines if the address in the parameter is an rfc1918 or rfc6598 address. The parameter allows pseudo-variables usage.
This function can be used from ANY_ROUTE.
Example 1.22. is_rfc1918 usage
...
if(is_rfc1918("$rd")) {
# domain in r-uri is private address
}
...
Adds an “;alias=ip~port~transport” parameter to the contact URI containing either received ip, port, and transport protocol or those given as parameters. If called without parameters, “;alias” parameter is only added if received ip, port, or transport protocol differs from that in contact URI.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE, and LOCAL_ROUTE.
Example 1.23. add_contact_alias usage
...
if (!is_present_hf("Record-Route")) {
if (!add_contact_alias("$var(src_ip)", "$Rp", "tcp")) {
xlog("L_ERR", "Error in aliasing contact $ct\n");
send_reply("400", "Bad request");
exit;
};
};
...
Checks if the Request URI has an “alias”
parameter and if so, removes it and sets the “$du” based
on its value. Note that this means that routing of request is based on
“;alias” parameter value of the Request URI rather than
the Request URI itself. If you call handle_ruri_alias()
on a request, make sure that you screen the alias parameter value of
Request URI the same way as you would screen the
Request URI itself.
Returns 1 if “;alias” parameter was found and “$du” was set and the “$ru” rewritten, 2 if the alias parameter was not found and nothing was done, or -1 in case of error.
This function can be used from REQUEST_ROUTE, BRANCH_ROUTE, and LOCAL_ROUTE.
Example 1.24. handle_ruri_alias usage
...
if ($du == "") {
handle_ruri_alias();
switch ($rc) {
case -1:
xlog("L_ERR", "Failed to handle alias of R-URI $ru\n");
send_reply("400", "Bad request");
exit;
case 1:
xlog("L_INFO", "Routing in-dialog $rm from $fu to $du\n");
break;
case 2:
xlog("L_INFO", "Routing in-dialog $rm from $fu to $ru\n");
break;
};
};
...
Adds an “;alias=ip~port~transport” parameter to the
contact URI containing the received ip, port, and transport protocol.
The update of contact URI is signaled to a few other modules in the
way the fix_nated_contact() does it by using the
internal flags. The new value is not visible to pseudo-variables and it
does not change the SIP message buffer.
Meaning of parameters:
trim - by default, set_contact_alias() will not detect and trim an already existing alias parameter. If this optional parameter is set to "1", set_contact_alias() will trim the existing alias before adding a new one.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE, and FAILURE_ROUTE.
Example 1.25. set_contact_alias usage
...
if (!is_present_hf("Record-Route")) {
if (!set_contact_alias()) {
xlog("L_ERR", "Error in aliasing contact $ct\n");
send_reply("400", "Bad request");
exit;
};
};
...
Reads “;alias=ip~port~transport” from Contact header then writes to target pseudo-variable as a sip uri.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE, and FAILURE_ROUTE.
Number of Record Routes in received SIP request or reply.
If topmost Record Route in received SIP request or reply is a double Record Route, value of $rr_top_count is 2. If it a single Record Route, value of $rr_top_count is 1. If there is no Record Route(s), value of $rr_top_count is 0.
Example 1.28. $rr_top_count usage
...
if ($rr_count == $avp(rr_count) + $rr_top_count) {
route(ADD_CONTACT_ALIAS);
};
...
Get n-th Contact value with IP:Port rewritten to source ip:port. N is counted from 1. Only IP:port is rewritten, remaining part are left unchanged. Full nameaddr is supported.
Example 1.30. @nathelper.rewrite_contact usage
... $c = @nathelper.rewrite_contact[1]; ... $c2 = @nathelper.rewrite_contact[1].nameaddr.uri;
|
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 Kamailio? |
|
Take a look at https://www.kamailio.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 Kamailio release should be sent to
|
|
|
2.4. |
How can I report a bug? |
|
Please follow the guidelines provided at: https://github.com/kamailio/kamailio/issues. |