Copyright © 2003-2008 Sippy Software, Inc.
Copyright © 2005 voice-system
Copyright © 2009 TuTPro Inc.
| Revision History | |
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| Revision $Revision$ | $Date$ |
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_tout (integer)
rtpproxy_retr (integer)
force_socket (string)
sipping_bflag (integer)
sipping_from (string)
sipping_method (string)
nortpproxy_str (string)
fix_nated_contact()
fix_nated_sdp(flags [, ip_address])
set_rtp_proxy_set()
force_rtp_proxy([flags [, ip_address]])
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()
add_contact_alias()
handle_ruri_alias()
List of Examples
natping_interval parameterping_nated_only parameternatping_processes parameternatping_socket parameterreceived_avp parameterrtpproxy_sock parameterrtpproxy_disable_tout parameterrtpproxy_tout parameterrtpproxy_retr parameterforce_socket parametersipping_bflag parametersipping_from parametersipping_method parameternortpproxy_str parameterfix_nated_contact usagefix_nated_sdp usagefix_nated_contact usageforce_rtp_proxy usagertpproxy_offer usageunforce_rtp_proxy usagertpproxy_stream2xxx usageadd_rcv_paramer usagefix_nated_register usagestart_recording usageadd_contact_alias usagehandle_ruri_alias usagenh_enable_ping usagenh_enable_rtpp usagenh_show_rtpp usageTable 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_tout (integer)
rtpproxy_retr (integer)
force_socket (string)
sipping_bflag (integer)
sipping_from (string)
sipping_method (string)
nortpproxy_str (string)
fix_nated_contact()
fix_nated_sdp(flags [, ip_address])
set_rtp_proxy_set()
force_rtp_proxy([flags [, ip_address]])
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()
add_contact_alias()
handle_ruri_alias()
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 indication to SDP (flag 0x01) and updates source IP address too (flag 0x02).
Alternative to fix_nated_contact() is add_contact_alias() that together with handle_ruri_alias() also supports reuse of tcp 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;
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 multiple sets, take care and use the same set for both force_ and unforce_rtpproxy()!!
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).
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")
...
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”.
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")
...
Rewrites Contact HF to contain request's source address:port.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE.
Example 1.15. 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 1.16. 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.
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 1.18. 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");};
...
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 1.19. 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. 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.
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,
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 1.22. 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();
};
};
...
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 1.23. 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
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.
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.
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 1.26. 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 $du based on its value. Note that this means that routing of request is based on alias parameter value of Request URI rather than Request URI itself. If you call handle_ruri_alias() on a request, make thus sure that you screen alias parameter value of Request URI the same way as you would screen Request URI itself.
This function can be used from REQUEST_ROUTE, BRANCH_ROUTE, and LOCAL_ROUTE.
Example 1.27. handle_ruri_alias usage
...
if ($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 1.29. $rr_top_count usage
...
if ($rr_count == $avp(rr_count) + $rr_top_count) {
route(ADD_CONTACT_ALIAS);
};
...
Enables natping if parameter value greater than 0. Disables natping if parameter value is 0.
The function takes only one parameter - a number in decimal format.
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.
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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. |