Copyright © 2003-2008 Sippy Software, Inc.
Copyright © 2005 Voice Sistem SRL
| Revision History | |
|---|---|
| Revision $Revision: 8740 $ | $Date: 2013-01-29 13:35:11 +0100 (Tue, 29 Jan 2013) $ |
Table of Contents
rtpproxy_sock (string)rtpproxy_disable_tout (integer)rtpproxy_timeout (string)rtpproxy_autobridge (integer)rtpproxy_tout (integer)rtpproxy_retr (integer)nortpproxy_str (string)db_url (string)db_table (string)rtpp_socket_col (string)set_id_col (string)rtpp_notify_socket (string)set_rtp_proxy_set()
engage_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()
start_recording()
List of Examples
rtpproxy_sock parameterrtpproxy_disable_tout parameterrtpproxy_timeout parameter to 200msrtpproxy_retr parameternortpproxy_str parameterdb_url parameterdb_table parameterrtpp_socket_col parameterset_id parameterrtpp_notify_socket parameterfix_nated_contact usageengage_rtp_proxy usagertpproxy_offer usagertpproxy_answer usageunforce_rtp_proxy usagertpproxy_stream2xxx usagestart_recording usagertpproxy_enable usagertpproxy_show usagertpproxy_reload usageThis module is used by OpenSIPS to communicate with RTPProxy, a media relay proxy used to make the communication between user agents behind NAT possible.
This module is also used along with RTPProxy to record media streams between user agents or to play media to either UAc or UAs.
Currently, the rtpproxy 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 rtpproxy 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 engage_rtp_proxy(), 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:
a database module - only if you want to load use a database table from where to load the rtp proxies sets.
dialog module - if using the engage_rtp_proxy functions or RTPProxy timeout notifications.
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 1.1. Set rtpproxy_sock parameter
...
# single rtproxy with specific weight
modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:12221=2")
# multiple rtproxies for LB
modparam("rtpproxy", "rtpproxy_sock",
"udp:localhost:12221 udp:localhost:12222")
# multiple sets of multiple rtproxies
modparam("rtpproxy", "rtpproxy_sock",
"1 == udp:localhost:12221 udp:localhost:12222")
modparam("rtpproxy", "rtpproxy_sock",
"2 == udp:localhost:12225")
...
Once RTPProxy was found unreachable and marked as disable, rtpproxy will not attempt to establish communication to RTPProxy for rtpproxy_disable_tout seconds.
Default value is “60”.
Example 1.2. Set rtpproxy_disable_tout parameter
...
modparam("rtpproxy", "rtpproxy_disable_tout", 20)
...
Timeout value in waiting for reply from RTPProxy.
Default value is “1”.
Example 1.3. Set rtpproxy_timeout parameter to 200ms
...
modparam("rtpproxy", "rtpproxy_timeout", "0.2")
...
Enable auto-bridging feature. Does not properly function when doing serial/parallel forking!
Default value is “0”.
How many times rtpproxy should retry to send and receive after timeout was generated.
Default value is “5”.
The parameter sets the SDP attribute used by rtpproxy 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 1.6. Set nortpproxy_str parameter
...
modparam("rtpproxy", "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 rtpproxy_reload is issued(the definitions will be replaced with the ones from the database table).
Default value is “NULL”.
Example 1.7. Set db_url parameter
...
modparam("rtpproxy", "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 “rtpproxy_sockets”.
The name rtpp socket column in the database table.
Default value is “rtpproxy_sock”.
Example 1.9. Set rtpp_socket_col parameter
...
modparam("rtpproxy", "rtpp_socket_col", "rtpp_socket")
...
The name set id column in the database table.
Default value is “set_id”.
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, FAILURE_ROUTE and LOCAL_ROUTE.
Rewrites SDP body to ensure that media is passed through an RTP proxy. It uses the dialog module facilities to keep track when the rtpproxy session must be updated. Function must only be called for the initial INVITE and internally takes care of rewriting the body of 200 OKs and ACKs. Note that when used in bridge mode, this function might advertise wrong interfaces in SDP (due to the fact that OpenSIPS is not aware of the RTPProxy configuration), so you might face an undefined behavior.
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/e - when RTPProxy is used in bridge mode, these flags are used to indicate the direction of the media flow for the current request/reply. 'i' refers to the LAN (internal network) and corresponds to the first interface of RTPProxy (as specified by the -l parameter). 'e' refers to the WAN (external network) and corresponds to the second interface of RTPProxy. These flags should always be used together. For example, an INVITE (offer) that comes from the Internet (WAN) to goes to a local media server (LAN) should use the 'ei' flags. The answer should use the 'ie' flags. Depending on the scenario, the 'ii' and 'ee' combination are also supported. Only makes sense when RTPProxy is running in the bridge mode.
f - instructs rtpproxy to ignore marks inserted by another rtpproxy 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, rtpproxy 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/w - flags that for the UA from which message is received, support symmetric RTP must be forced.
n - flags that enables the notification timeout for the session.
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.
Both parmeters do accept variables in any combination.
This function can be used from REQUEST_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE.
Example 1.13. engage_rtp_proxy usage
...
if (is_method("INVITE") &&
has_totag()) {engage_rtp_proxy();};
...
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.
See engage_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 1.14. 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 engage_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
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 1.17. 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.
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 1.19.
rtpproxy_enable usage
... $ opensipsctl fifo rtpproxy_enable 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. |