Copyright © 2003-2008 Sippy Software, Inc.
Copyright © 2005 voice-system
Copyright © 2009 TuTPro Inc.
Copyright © 2010 VoIPEmbedded Inc.
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Table of Contents
List of Examples
natping_interval parameterping_nated_only parameternatping_processes parameternatping_socket parameterreceived_avp parametersipping_bflag parametersipping_from parametersipping_method parameternortpproxy_str parameterfix_nated_contact usagefix_nated_sdp usagefix_nated_contact usageadd_rcv_paramer usagefix_nated_register usageadd_contact_alias usagehandle_ruri_alias usagenh_enable_ping usageTable of Contents
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.
Function fix_nated_contact() rewrites Contact header field with request's source address:port pair. Function fix_nated_sdp() adds the active direction ndication to SDP (flag 0x01) and updates source IP address too (flag 0x02). 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 handle_ruri_alias() is standards conforming and also supports reuse of TCP/TLS connections.
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">
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 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 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 bandwitdh 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.
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).
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 1.7. 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 1.9. Set nortpproxy_str parameter
...
modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n")
...
Rewrites Contact HF to contain request's source address:port.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE.
Example 1.10. fix_nated_contact usage
...
if (search("User-Agent: Cisco ATA.*") {fix_nated_contact();};
...
Alters the SDP information in orer to facilitate NAT traversal. What changes to be performed may be controled via the “flags” parameter.
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 1.11. 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 [un]force_rtp_proxy(), rtpproxy_offer() or rtpproxy_answer() command.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, BRANCH_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 1.13. 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 cloumn 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 - 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 - test if the source IP address of signaling is a RFC1918 address
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.
Adds ;alias=ip:port parameter to contact URI containing received ip:port if contact uri ip:port does not match received ip:port.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE, and LOCAL_ROUTE.
Example 1.15. add_contact_alias usage
...
if (!is_present_hf("Record-Route")) {
if (!add_contact_alias()) {
xlog("L_ERR", "Error in aliasing contact $ct\n");
send_reply("400", "Bad request");
exit;
};
};
...
Checks if Request URI has alias param and if so, removes it and sets $du based on its value. Note that this means that routing of request is based on ;alias parameter value of Request URI rather than Request URI itself. If you call handle_ruri_alias() on a request, make thus sure that you screen alias parameter value of Request URI the same way as you would screen Request URI itself.
Returns 1 if ;alias param was found and $du was set and $ru rewritten, 2 if alias param was not found and nothing was done, or -1 in case of error.
This function can be used from REQUEST_ROUTE, BRANCH_ROUTE, and LOCAL_ROUTE.
Example 1.16. handle_ruri_alias usage
...
if ($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;
};
};
...
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 1.18. $rr_top_count usage
...
if ($rr_count == $avp(rr_count) + $rr_top_count) {
route(ADD_CONTACT_ALIAS);
};
...
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2.1. |
What happend with “rtpproxy_disable” parameter? |
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It was removed as it became obsolete - now “rtpproxy_sock” can take empty value to disable the rtpproxy functionality. |
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2.2. |
Where can I find more about Kamailio? |
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Take a look at http://www.kamailio.org/. |
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2.3. |
Where can I post a question about this module? |
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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
If you want to keep the mail private, send it to
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2.4. |
How can I report a bug? |
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Please follow the guidelines provided at: http://sourceforge.net/tracker/?group_id=139143. |