Table of Contents
rtpproxy_sock (string)rtpproxy_disable_tout (integer)rtpproxy_timeout (string)rtpproxy_autobridge (integer)rtpproxy_tout (integer)rtpproxy_retr (integer)default_set (integer)nortpproxy_str (string)db_url (string)db_table (string)rtpp_socket_col (string)set_id_col (string)rtpp_notify_socket (string)rtpproxy_engage([[flags][, [ip_address][, [set_id][, [sock_pvar][, ret_pvar]]]]])
rtpproxy_offer([[flags][, [ip_address][, [set_id][, [sock_pvar][, ret_pvar]]]])
rtpproxy_answer([[flags][, [ip_address][, [set_id][, [sock_pvar][, ret_pvar]]]]])
rtpproxy_unforce([[set_id][, sock_pvar]])
rtpproxy_stream2uac(prompt_name, count[, [set_id][, sock_pvar]]),
rtpproxy_stream2uas(prompt_name, count[, [set_id][, sock_pvar]])
rtpproxy_stop_stream2uac([[set_id][, sock_pvar]]),
rtpproxy_stop_stream2uas([[set_id][, sock_pvar]])
rtpproxy_start_recording([[set_id][, [sock_pvar][, [flags][, [destination][, mediastream]]]]])
rtpproxy_stats(up_pvar, down_pvar, sent_pvar, fail_pvar[, [set_id][, sock_pvar]])
rtpproxy_all_stats(stats_avp[, [set_id][, sock_pvar]])
List of Tables
List of Examples
rtpproxy_sock parameterrtpproxy_disable_tout parameterrtpproxy_timeout parameter to 200msrtpproxy_retr parameterdefault_set parameternortpproxy_str parameterdb_url parameterdb_table parameterrtpp_socket_col parameterset_id parameterrtpp_notify_socket parameterrtpproxy_engage usagertpproxy_offer usagertpproxy_answer usagertpproxy_unforce usagertpproxy_stream2xxx usagertpproxy_start_recording usagertpproxy_stats usagertpproxy_all_stats 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.
Starting with OpenSIPS 2.1, engage_rtp_proxy(), unforce_rtp_proxy() and start_recording() functions have been fully replaced by rtpproxy_engage(), rtpproxy_unforce() and rtpproxy_start_recording().
IMPORTANT: if you use multiple sets, make sure you use the same set for both rtpproxy_offer()/rtpproxy_answer() and rtpproxy_unforce()!!
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.
Note that RTPProxy version v2.0.0 has integrated this feature upstream, therefore this patch is no longer needed.
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_engage(), 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: https://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 rtpproxy_engage 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 indicates the default RTPProxy set to be used when provisioning an engine in the config file without an explicit set, or when calling one of the rtpproxy_*() functions without an explicit set.
Default value is set “0”.
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 1.7. 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 1.8. 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 1.10. 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”.
The socket used by OpenSIPS to receive timeout notifications.
Default value is “NULL”.
Example 1.12. Set rtpp_notify_socket parameter
...
modparam("rtpproxy", "rtpp_notify_socket", "tcp:10.10.10.10:9999")
# use an UNIX socket
modparam("rtpproxy", "rtpp_notify_socket", "unix:/tmp/rtpproxy.unix")
# or
modparam("rtpproxy", "rtpp_notify_socket", "/tmp/rtpproxy.unix")
...
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(optional) - 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.
NOTE: when using RTPProxy in bridge mode, all sessions are considered asymmetric (as oposed to symmetric if used in normal mode). If you have symmetric clients (this is the most common scenario), you'll have to force the s!
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.
tNN - can be used to specify a RTP ttl for the caller. The NN represents the timeout in seconds for that stream. This can be useful in music on hold scenarios where only one client is sending RTP.
TNN - Similar to the tNN paramaeter, but used for tuning the calllee's ttl for RTP.
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(optional) - new SDP IP address.
set_id(optional) - the set used for this call.
sock_pvar(optional) - pvar used to store the RTPProxy socket chosen for this call. Note that the variable will only be populated in the initial request.
ret_pvar(optional) - pvar used to print the IP and port the RTPProxy server is using for this call. This is useful especially when using the rtp_cluster, which can advertise multiple servers behind it. The format of the value returned is IP:port. Note that the variable will only be populated in the initial request.
This function can be used from REQUEST_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE.
Example 1.13. rtpproxy_engage usage
...
if (is_method("INVITE") && has_totag()) {
if ($var(setid) != 0) {
rtpproxy_engage(,,"$var(setid)", "$var(proxy)");
xlog("SCRIPT: RTPProxy server used is $var(proxy)\n");
} else {
rtpproxy_engage();
xlog("SCRIPT: using default RTPProxy set\n");
}
}
...
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 rtpproxy_engage() 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_body("application/sdp")) {
if (rtpproxy_offer())
t_on_reply("1");
} else {
t_on_reply("2");
}
}
if (is_method("ACK") && has_body("application/sdp"))
rtpproxy_answer();
...
}
onreply_route[1]
{
...
if (has_body("application/sdp"))
rtpproxy_answer();
...
}
onreply_route[2]
{
...
if (has_body("application/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_engage() 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.
Meaning of the parameters is as follows:
set_id(optional) - the set used for this call.
sock_pvar(optional) - pvar used to store the RTPProxy socket chosen for this 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.
set_id(optional) - the set used for this call.
sock_pvar(optional) - pvar used to store the RTPProxy socket chosen for this call.
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.
Meaning of the parameters is as follows:
set_id(optional) - the set used for this call.
sock_pvar(optional) - pvar used to store the RTPProxy socket chosen for this call.
These functions can be used from REQUEST_ROUTE, ONREPLY_ROUTE.
This command will send a signal to the RTP-Proxy to record the RTP stream on the RTP-Proxy.
Meaning of the parameters is as follows:
set_id(optional) - the set used for this call.
sock_pvar(optional) - pvar used to store the RTPProxy socket chosen for this call.
flags(optional) - a list of flags passed to RTPProxy for the recording. Currently only s is supported, and it indicates that RTPProxy should record both audio legs in a single file. Note that this feature is available starting with RTPProxy 2.0.
destination(optional) - the destination of the recording. If it has the udp:IP:port format, RTPProxy sends the RTP stream to that IP:port remote destination. Otherwise, destination represents the name of the file in the recording directory.
mediastream(optional) - this parameter is only used if the destination is specified, and represents the index of media stream to record/copy, starting from 1. If this parameter is missing, OpenSIPS instructs RTPProxy to copy all the streams.
This function can be used from REQUEST_ROUTE and ONREPLY_ROUTE.
Example 1.18. rtpproxy_start_recording usage
... rtpproxy_start_recording(); # copy RTP stream to a different listener rtpproxy_start_recording(,,,"udp:127.0.0.1:60000"); # copy only first RTP stream (audio stream) rtpproxy_start_recording(,,,"udp:127.0.0.1:60000", "1"); ...
This command gathers call RTP statistics from RTP-Proxy.
Meaning of the parameters is as follows:
up_pvar - the pseudo-variable used to return the packets sent by upstream for this call.
down_pvar - the pseudo-variable used to return the packets sent by downstream for this call.
sent_pvar - the pseudo-variable used to return the total number of packets sent for this call.
up_pvar - the pseudo-variable used to return the number of failed packets for this call.
set_id(optional) - the set used for this call.
sock_pvar(optional) - pvar used to store the RTPProxy socket chosen for this call.
This function can be used from REQUEST_ROUTE, FAILURE_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE and LOCAL_ROUTE.
Example 1.19. rtpproxy_stats usage
...
rtpproxy_stats("$var(up)","$var(down)","$var(sent)","$var(fail)");
xlog("RTP statistics for $ci: up=$var(up) down=$var(down) sent=$var(sent) fail=$var(fail)\n");
...
This command gathers all RTP statistics available from RTP-Proxy. All the returned values stored in an AVP that can be further read by indexing the AVP.
This command is only available starting with RTPProxy 2.1 realease.
Meaning of the parameters is as follows:
stats_avp - an AVP where the statistics will be stored. This AVP can be further indexed to get a specific statistic.
set_id(optional) - the set used for this call.
sock_pvar(optional) - pvar used to store the RTPProxy socket chosen for this call.
This function can be used from REQUEST_ROUTE, FAILURE_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE and LOCAL_ROUTE.
Each statistic is stored at a specific index as it follows:
ttl - $avp(ret) / $(avp(ret)[0])
pkts_ia - $(avp(ret)[1])
pkts_io - $(avp(ret)[2])
relayed - $(avp(ret)[3])
dropped - $(avp(ret)[4])
rtpa_set - $(avp(ret)[5])
rtpa_rcvd - $(avp(ret)[6])
rtpa_dups - $(avp(ret)[7])
rtpa_lost - $(avp(ret)[8])
rtpa_perrs - $(avp(ret)[9])
Example 1.20. rtpproxy_all_stats usage
...
rtpproxy_all_stats("$avp(stats)");
xlog("RTP statistics for $ci: dropped=$(avp(stats)[4])\n");
...
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 next parameter (optional) is the rtpproxy set ID (used for better indentification of the rtpproxy instance to be enabled, for example when a rtpproxy is used in multiple sets).
The last parameter must be a number in decimal representing the new enabled/disabled state.
NOTE: if a rtpproxy is defined multiple times (in the same or different set), all its instances will be enables/disabled IF no set ID provided (as second param).
Example 1.21.
rtpproxy_enable usage
... ## disable a RTPProxy by URL only $ opensipsctl fifo rtpproxy_enable udp:192.168.2.133:8081 0 ## disable a RTPProxy by URL and set ID (3) $ opensipsctl fifo rtpproxy_enable udp:192.168.2.133:8081 3 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 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 OpenSIPS? |
Take a look at https://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. |
Table 3.1. Top contributors by DevScore(1), authored commits(2) and lines added/removed(3)
| Name | DevScore | Commits | Lines ++ | Lines -- | |
|---|---|---|---|---|---|
| 1. | Razvan Crainea (@razvancrainea) | 153 | 100 | 3206 | 1457 |
| 2. | Maksym Sobolyev (@sobomax) | 40 | 3 | 4527 | 3 |
| 3. | Bogdan-Andrei Iancu (@bogdan-iancu) | 25 | 22 | 119 | 109 |
| 4. | Liviu Chircu (@liviuchircu) | 21 | 16 | 91 | 183 |
| 5. | Ovidiu Sas (@ovidiusas) | 7 | 5 | 31 | 22 |
| 6. | Vlad Paiu (@vladpaiu) | 6 | 4 | 15 | 9 |
| 7. | Peter Lemenkov (@lemenkov) | 4 | 2 | 59 | 8 |
| 8. | Vlad Patrascu (@rvlad-patrascu) | 4 | 2 | 25 | 32 |
| 9. | John Burke | 4 | 2 | 8 | 6 |
| 10. | Dave Sidwell (@davesidwell) | 4 | 2 | 8 | 2 |
All remaining contributors: Ezequiel Lovelle (@lovelle), Ryan Bullock (@rrb3942), Christophe Sollet (@csollet), Anca Vamanu, Mikko Lehto, Dan Pascu (@danpascu), Walter Doekes (@wdoekes), Dusan Klinec (@ph4r05), Julián Moreno Patińo, Zero King (@l2dy).
(1) DevScore = author_commits + author_lines_added / (project_lines_added / project_commits) + author_lines_deleted / (project_lines_deleted / project_commits)
(2) including any documentation-related commits, excluding merge commits. Regarding imported patches/code, we do our best to count the work on behalf of the proper owner, as per the "fix_authors" and "mod_renames" arrays in opensips/doc/build-contrib.sh. If you identify any patches/commits which do not get properly attributed to you, please submit a pull request which extends "fix_authors" and/or "mod_renames".
(3) ignoring whitespace edits, renamed files and auto-generated files
Table 3.2. Most recently active contributors(1) to this module
| Name | Commit Activity | |
|---|---|---|
| 1. | John Burke | Apr 2021 - Apr 2021 |
| 2. | Razvan Crainea (@razvancrainea) | Mar 2011 - Apr 2021 |
| 3. | Ovidiu Sas (@ovidiusas) | Mar 2011 - Jun 2020 |
| 4. | Bogdan-Andrei Iancu (@bogdan-iancu) | Mar 2011 - Jun 2020 |
| 5. | Zero King (@l2dy) | Mar 2020 - Mar 2020 |
| 6. | Dan Pascu (@danpascu) | May 2019 - May 2019 |
| 7. | Liviu Chircu (@liviuchircu) | Jul 2012 - Nov 2018 |
| 8. | Vlad Patrascu (@rvlad-patrascu) | May 2017 - Aug 2017 |
| 9. | Maksym Sobolyev (@sobomax) | Mar 2011 - Jul 2017 |
| 10. | Julián Moreno Patińo | Feb 2016 - Feb 2016 |
All remaining contributors: Vlad Paiu (@vladpaiu), Dusan Klinec (@ph4r05), Dave Sidwell (@davesidwell), Ezequiel Lovelle (@lovelle), Mikko Lehto, Ryan Bullock (@rrb3942), Peter Lemenkov (@lemenkov), Walter Doekes (@wdoekes), Christophe Sollet (@csollet), Anca Vamanu.
(1) including any documentation-related commits, excluding merge commits
Last edited by: Zero King (@l2dy), Liviu Chircu (@liviuchircu), Bogdan-Andrei Iancu (@bogdan-iancu), Razvan Crainea (@razvancrainea), Julián Moreno Patińo, Mikko Lehto, Ryan Bullock (@rrb3942), Maksym Sobolyev (@sobomax), Ovidiu Sas (@ovidiusas).
doc copyrights:
Copyright © 2005 Voice Sistem SRL
Copyright © 2003-2008 Sippy Software, Inc.