Table of Contents
rtpengine_sock (string)rtpengine_disable_tout (integer)rtpengine_tout (integer)rtpengine_retr (integer)rtpengine_timer_interval (integer)notification_sock (string)extra_id_pv (string)setid_avp (string)error_pv (string)db_url (string)db_table (string)socket_column (string)set_column (string)rtpengine_use_set(setid)
rtpengine_offer([flags[, sock_var[, sdp_pvar[, body]]]])
rtpengine_answer([flags[, sock_pvar[, sdp_pvar[, body]]]])
rtpengine_delete([flags[, sock_var]])
rtpengine_manage([flags[, sock_var[, sdp_var[, body]]]])
rtpengine_start_recording([flags [, sock_var]])
rtpengine_stop_recording([flags [, sock_var]])
rtpengine_play_media(flags, [duration_spec[, sock_var[, sockvar]]])
rtpengine_stop_media([flags[, sockvar]])
rtpengine_block_media([flags[, sockvar]])
rtpengine_unblock_media([flags[, sockvar]])
rtpengine_block_dtmf([flags[, sockvar]])
rtpengine_unblock_dtmf([flags[, sockvar]])
rtpengine_start_forwarding([flags[, sockvar]])
rtpengine_stop_forwarding([flags[, sockvar]])
rtpengine_play_dtmf(code, [flags[, sockvar]])
List of Tables
List of Examples
rtpengine_sock parameterrtpengine_disable_tout parameterrtpengine_tout parameterrtpengine_retr parameterrtpengine_timer_interval parameternotification_sock parameterextra_id_pv parametersetid_avp parametererror_pv parameterdb_url parameterdb_table parametersocket_column parameterset_column parameterrtpengine_use_set usagertpengine_offer usagertpengine_offer usage with body replacertpengine_offer usage with call recordingrtpengine_offer usage for transcodingrtpengine_answer usagertpengine_delete usagertpengine_manage usagertpengine_start_recording usagertpengine_stop_recording usagertpengine_play_mediartpengine_play_mediartpengine_stop_mediartpengine_block_media usagertpengine_unblock_media usagertpengine_block_dtmf usagertpengine_unblock_dtmf usagertpengine_start_forwarding usagertpengine_stop_forwarding usagertpengine_play_dtmf usagertpengine_enable usagertpengine_show usagertpengine_reload usageteardown usageThis is a module that enables media streams to be proxied via an RTP proxy. The only RTP proxy currently known to work with this module is the Sipwise rtpengine https://github.com/sipwise/rtpengine. The rtpengine module is a modified version of the original rtpproxy module using a new control protocol. The module is designed to be a drop-in replacement for the old module from a configuration file point of view, however due to the incompatible control protocol, it only works with RTP proxies which specifically support it.
The rtpengine module can support multiple RTP proxies for balancing/distribution and control/selection purposes.
The module allows definition of several sets of rtpengines. Load-balancing will be performed over a set and the admin has the ability to choose what set should be used. The set is selected via its id - the id being defined with the set. Refer to the “rtpengine_sock” module parameter definition for syntax description.
The balancing inside a set is done automatically by the module based on the weight of each RTP proxy from the set.
The selection of the set is done from script prior using rtpengine_delete(), rtpengine_offer() or rtpengine_answer() functions - see the rtpengine_use_set() function.
Another way to select the set is to define setid_avp module parameter and assign setid to the defined avp before calling rtpengine_offer() or rtpengine_manage() function. If forwarding of the requests fails and there is another branch to try, remember to unset the avp after calling rtpengine_delete() function.
For backward compatibility reasons, a set with no id take by default the id 0. Also if no set is explicitly set before rtpengine_delete(), rtpengine_offer() or rtpengine_answer() the 0 id set will be used.
IMPORTANT: if you use multiple sets, take care and use the same set for both rtpengine_offer()/rtpengine_answer() and rtpengine_delete()!! If the set was selected using setid_avp, the avp needs to be set only once before rtpengine_offer() or rtpengine_manage() call.
The following modules must be loaded before this module:
tm module - (optional) if you want to have rtpengine_manage() fully functional
Definition of socket(s) used to connect to (a set) RTP proxy. It may specify a UNIX socket or an IPv4/IPv6 UDP socket. If the protocol part (i.e. “udp:”) is missing, the socket is treated as a UNIX socket.
Default value is “NONE” (disabled).
Example 1.1. Set rtpengine_sock parameter
...
# single rtproxy
modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221")
# multiple rtproxies for LB
modparam("rtpengine", "rtpengine_sock",
"udp:localhost:12221 udp:localhost:12222")
# multiple sets of multiple rtproxies
modparam("rtpengine", "rtpengine_sock",
"1 == udp:localhost:12221 udp:localhost:12222")
modparam("rtpengine", "rtpengine_sock",
"2 == udp:localhost:12225")
...
Once an RTP proxy was found unreachable and marked as disabled, the rtpengine module will not attempt to establish communication to that RTP proxy for rtpengine_disable_tout seconds.
Default value is “60”.
Example 1.2. Set rtpengine_disable_tout parameter
...
modparam("rtpengine", "rtpengine_disable_tout", 20)
...
Timeout value in waiting for reply from RTP proxy.
Default value is “1”.
How many times the module should retry to send and receive after timeout was generated.
Default value is “5”.
Frequency to scan rtpengine sets for disabled node probing. Probing is done outside the SIP processing context and in a separate timer routine. Disabled nodes are probed for re-enablement after rtpengine_disable_tout seconds. Setting this value too high can lead to unexpectedly large disabled interval as the max interval before probing is (rtpengine_timer_interval + rtpengine_disable_tout) seconds.
Default value is “5”.
Example 1.5. Set rtpengine_timer_interval parameter
...
modparam("rtpengine", "rtpengine_timer_interval", 1)
...
An UDP socket formatted as IP:port that indicates the listening IP and port OpenSIPS will bind for to receive notifications (such as DTMF events) from RTPengine.
Every notification received from RTPengine will trigger an E_RTPENGINE_NOTIFICATION event.
Default value is “none” - notifications are ignored.
Example 1.6. Set notification_sock parameter
...
modparam("rtpengine", "notification_sock", "127.0.0.1:9999")
...
The parameter sets the PV definition to use when the “via-branch=extra” option is used on the rtpengine_delete(), rtpengine_offer(), rtpengine_answer() or rtpengine_manage() commands.
Default is empty, the “via-branch=extra” option may not be used then.
Example 1.7. Set extra_id_pv parameter
...
modparam("rtpengine", "extra_id_pv", "$avp(extra_id)")
...
The parameter defines an AVP that, if set, determines which RTP proxy set rtpengine_offer(), rtpengine_answer(), rtpengine_delete(), and rtpengine_manage() functions use.
There is no default value.
The parameter defines a variable that shall be populated by RTP when one of the rtpengine_* functions fail.
There is no default value.
Example 1.9. Set error_pv parameter
...
modparam("rtpengine", "error_pv", "$var(rtpengine_error)")
...
Database URL, used to load RTPEngines sockets from db, instead of specifying them in the script (rtpengine_sock module parameter).
Default value is “NULL”, no database is used.
Example 1.10. Set db_url parameter
...
modparam("rtpengine", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
...
The table where the RTPEngines sockets are stored. Used when Database URL is provisioned.
Default value is “rtpengine”.
The name of the rtpengine socket column in the database table.
Default value is “socket”.
Sets the ID of the RTP proxy set to be used for the next rtpengine_delete(), rtpengine_offer(), rtpengine_answer() or rtpengine_manage() command. The parameter is an integer.
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. 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.
Meaning of the parameters is as follows:
flags(string, optional) - flags to turn on some features.
The “flags” string is a list of space-separated items. Each item is either an individual token, or a token in “key=value” format. The possible tokens are described below.
When passing an option that OpenSIPS is not aware of, it will be blindly sent to the rtpengine daemon to be processed.
via-branch=... - Include the “branch” value of one of the “Via” headers in the request to the RTP proxy. Possible values are: “1” - use the first “Via” header; “2” - use the second “Via” header; “auto” - use the first “Via” header if this is a request, or the second one if this is a reply; “extra” - don't take the value from a header, but instead use the value of the “extra_id_pv” variable. This can be used to create one media session per branch on the RTP proxy. When sending a subsequent “delete” command to the RTP proxy, you can then stop just the session for a specific branch when passing the flag '1' or '2' in the “rtpengine_delete”, or stop all sessions for a call when not passing one of those two flags there. This is especially useful if you have serially forked call scenarios where the RTP proxy gets an “offer” command for a new branch, and then a “delete” command for the previous branch, which would otherwise delete the full call, breaking the subsequent “answer” for the new branch. This flag is only supported by the Sipwise rtpengine RTP proxy at the moment!
via-branch-param=... - provide a custom value for the via-branch param.
call-id - provide a custom Call-ID for the session. If missing, the Call-Id of the request/reply is used.
from-tag - provide a custom from-tag for the session. If missing, the from-tag request is used.
to-tag - provide a custom to-tag of the session. If missing, the to-tag of the request/reply is used, is present.
asymmetric - flags that UA from which message is received doesn't support symmetric RTP. (automatically sets the 'r' flag)
force-answer - force “answer”, that is, only rewrite SDP when corresponding session already exists in the RTP proxy. By default is on when the session is to be completed.
in-iface=..., out-iface=... - these flags specify the direction the SIP message. These flags only make sense when the RTP proxy is running in bridge mode. “in-iface” should indicate the proxy's inbound interface, and “out-iface” corresponds to the RTP proxy's outbound interface. You always have to specify two flags to define the incoming network and the outgoing network. For example, “in-iface=internal out-iface=external” should be used for SIP message received from the local interface and sent out on the external interface.
internal, external - these the old flags used to specify the direction of call. They are now obsolate, being replaced by the “in-iface=internal out-iface=external” configuration.
auto-bridge - this flag an alternative to the “internal” and “external” flags in order to do automatic bridging between IPv4 on the "internal network" and IPv6 on the "external network". Instead of explicitly instructing the RTP proxy to select a particular address family, the distinction is done by the given IP in the SDP body by the RTP proxy itself. Not supported by Sipwise rtpengine.
address-family=... - instructs the RTP proxy that the recipient of this SDP body expects to see addresses of a particular family. Possible values are “IP4” and “IP6”. For example, if the SDP body contains IPv4 addresses but the recipient only speaks IPv6, you would use “address-family=IP6” to bridge between the two address families.
Sipwise rtpengine remembers the address family preference of each party after it has seen an SDP body from them. This means that normally it is only necessary to explicitly specify the address family in the “offer”, but not in the “answer”.
Note: Please note, that this will only work properly with non-dual-stack user-agents or with dual-stack clients according to RFC6157 (which suggest ICE for Dual-Stack implementations). This short-cut will not work properly with RFC4091 (ANAT) compatible clients, which suggests having different m-lines with different IP-protocols grouped together.
received-from=... - sets the address from which SIP packet with SDP received. This flag always set automatically, don't use it until you have a reason for that.
force - instructs the RTP proxy to ignore marks inserted by another RTP proxy in transit to indicate that the session is already goes through another proxy. Allows creating a chain of proxies. Not supported and ignored by Sipwise rtpengine.
trust-address - flags that IP address in SDP should be trusted. Without this flag, the RTP proxy ignores address in the SDP and uses source address of the SIP message as media address which is passed to the RTP proxy. From rtpengine 3.8 this is the default behaviour.
SIP-source-address - the opposite of trust-address. Restores the old default behaviour of ignoring endppoint of the addresses in the SDP body.
replace-origin - flags that IP from the origin description (o=) should be also changed.
replace-session-connection - flags to change the session-level SDP connection (c=) IP if media description also includes connection information.
replace-zero-address - flags to replace zero address with real address. Using a zero endpoint address is an obsolete way to signal a muted or sendonly stream. Streams with zero addresses are normally flagged as sendonly and the zero address in the SDP is passed through.
symmetric - flags that for the UA from which message is received, support symmetric RTP must be forced. You do not need to explicitly specify this value, as it is the default, and the behavior is only changed when the asymmetric is used.
repacketize=NN - requests the RTP proxy 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 RTP proxy 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. Not supported by Sipwise rtpengine.
loop-protect - flag that instructs RTP to avoid rewriting the SDP when looping the same message.
ICE=... - controls the RTP proxy's behaviour regarding ICE attributes within the SDP body. Possible values are: “force” - discard any ICE attributes already present in the SDP body and then generate and insert new ICE data, leaving itself as the only ICE candidates; “remove” instructs the RTP proxy to discard any ICE attributes and not insert any new ones into the SDP. The default (if no “ICE=...” is given at all), new ICE data will only be generated if no ICE was present in the SDP originally; otherwise the RTP proxy will only insert itself as an additional ICE candidate. Other SDP substitutions (c=, m=, etc) are unaffected by this flag.
RTP, SRTP, AVP, AVPF - These flags control the RTP transport protocol that should be used towards the recipient of the SDP. If none of them are specified, the protocol given in the SDP is left untouched. Otherwise, the “SRTP” flag indicates that SRTP should be used, while “RTP” indicates that SRTP should not be used. “AVPF” indicates that the advanced RTCP profile with feedback messages should be used, and “AVP” indicates that the regular RTCP profile should be used. See also the next set of flags below.
RTP/AVP, RTP/SAVP, RTP/AVPF, RTP/SAVPF - these serve as an alternative, more explicit way to select between the different RTP protocols and profiles supported by the RTP proxy. For example, giving the flag “RTP/SAVPF” has the same effect as giving the two flags “SRTP AVPF”.
to-tag - force inclusion of the “To” tag. Normally, the “To” tag is always included when present, except for “delete” messages. Including the “To” tag in a “delete” messages allows you to be more selective about which dialogues within a call are being torn down.
to-tag=... - use the specified string as “To” tag instead of the actual “To” tag from the SIP message, and force inclusion of the tag in the message as per above.
from-tag=... - use the specified string as “From” tag instead of the actual “From” tag from the SIP message.
call-id=... - use the specified string as “Call-ID” instead of the actual “Call-ID” from the SIP message.
rtcp-mux-demux - if rtcp-mux (RFC 5761) was offered, make the RTP proxy accept the offer, but not offer it to the recipient of this message.
rtcp-mux-reject - if rtcp-mux was offered, make the RTP proxy reject the offer, but still offer it to the recipient. Can be combined with “rtcp-mux-offer” to always offer it.
rtcp-mux-offer - make the RTP proxy offer rtcp-mux to the recipient of this message, regardless of whether it was offered originally or not.
rtcp-mux-require - Similar to offer but pretends that the client has accepted rtcp-mux. This breaks RFC 5761 and will not advertise seperate RTCP ports. This option is necessary for WebRTC clients.
rtcp-mux-accept - if rtcp-mux was offered, make the RTP proxy accept the offer and also offer it to the recipient of this message. Can be combined with “rtcp-mux-offer” to always offer it.
media-address=... - force a particular media address to be used in the SDP body. Address family is detected automatically.
record-call=yes/no - indicates whether rtpengine should record the call or not. When using this parameter, you may pass further information in the “metadata”.
transcode-CODEC - used only for offer, indicates that rtpengine should transcode the CODEC towards the B-side. Example: transcode-PCMA will present to the B-side the PCMA codec.
codec-strip-CODEC - used only for offer, indicates that the A-side of the call will not end up talking CODEC. Example: codec-strip-PCMA will prevent the A-side from receiving the PCMA codec.
codec-mask-CODEC - used only for offer, indicates that the A-side will use the CODEC, but it will not be presented to the B-side. Example: codec-mask-PCMA will make the A-side receive the PCMA codec, but B-side will use something else.
sock_var(var, optional) - variable used to store the rtpengine socket chosen for this call.
sdp_var(var, optional) - variable used to store the full SDP received from rtpengine. You can perform any additional changes on this string. Important: when providing this variable, the message body is no longer changed, so you have to manually replace it!.
body(string, optional) - used to provide a specific body to the rtpengine_* function. If this parameter is missing the body of the current message is used.
This function can be used from ALL_ROUTES.
Example 1.15. rtpengine_offer usage
route {
...
if (is_method("INVITE")) {
if (has_body("application/sdp")) {
if (rtpengine_offer())
t_on_reply("1");
} else {
t_on_reply("2");
}
}
if (is_method("ACK") && has_body("application/sdp"))
rtpengine_answer();
...
}
onreply_route[1]
{
...
if (has_body("application/sdp"))
rtpengine_answer();
...
}
onreply_route[2]
{
...
if (has_body("application/sdp"))
rtpengine_offer();
...
}
Example 1.16. rtpengine_offer usage with body replace
...
if (rtpengine_offer(, $var(socket), $var(body), $rb)) {
xlog("Used rtpengine $var(socket)\n");
# make all the changes on the resulted SDP in $var(body)
...
remove_body_part();
add_body_part($var(body), "application/sdp");
}
...
Example 1.17. rtpengine_offer usage with call recording
...
$var(rtpengine_flags) = $var(rtpengine_flags) + " record-call=yes";
$json(recording_keys) := "{}";
$json(recording_keys/callId) = $ci;
$json(recording_keys/fromUser) = $dlg_val(recording_from_user);
$json(recording_keys/fromDomain) = $dlg_val(recording_from_domain);
$json(recording_keys/fromTag) = $dlg_val(recording_from_tag);
$json(recording_keys/toUser) = $dlg_val(recording_to_user);
$json(recording_keys/toDomain) = $dlg_val(recording_to_domain);
$var(rtpengine_flags) = $var(rtpengine_flags) + " metadata=" + $(json(recording_keys){s.encode.hexa});
rtpengine_offer($var(rtpengine_flags));
...
Example 1.18. rtpengine_offer usage for transcoding
...
# Goal: make A-side talk PCMA and B-side talk opus
# * do not present PCMA to B-side: codec-mask-PCMA, but use it on A-side
# * do not use opus for A-side: codec-strip-opus
# * offer opus to B-side: transcode-opus
rtpengine_offer("... codec-mask-PCMA codec-strip-opus transcode-opus ...");
...
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 rtpengine_offer() 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.
See rtpengine_offer() function description above for the meaning of the parameters. Note that not all flags make sense for a “delete”.
This function can be used from ALL_ROUTES.
Manage the RTPProxy session - it combines the functionality of rtpengine_offer(), rtpengine_answer() and rtpengine_delete(), detecting internally based on message type and method which one to execute.
It can take the same parameters as rtpengine_offer().
The flags parameter to rtpengine_manage() can be a configuration variable
containing the flags as a string.
Functionality:
If INVITE with SDP, then do rtpengine_offer()
If ACK with SDP, then do rtpengine_answer()
If BYE or CANCEL, or called within a FAILURE_ROUTE[], then do rtpengine_delete()
If reply to INVITE with code >= 300 do rtpengine_delete()
If reply with SDP to INVITE having code 1xx and 2xx, then
do rtpengine_answer() if the request had SDP or tm is not loaded,
otherwise do rtpengine_offer()
This function can be used from ALL_ROUTES.
This function will send a signal to the RTP proxy to record the RTP stream on the RTP proxy.
Meaning of the parameters is as follows:
flags(string, optional) - flags used to change the behavior of the recorder. An importat value to set is the call-id value, which can be used to start recording a different call than the requested one.
sock_var(var, optional) - variable used to store the rtpengine socket chosen for this call.
This function can be used from any route.
This function will send a signal to the RTP proxy to stop recording the RTP stream on the RTP proxy.
Meaning of the parameters is as follows:
flags(string, optional) - flags used to change the behavior of the recorder. An importat value to set is the call-id value, which can be used to start recording a different call than the requested one.
sock_var(var, optional) - variable used to store the rtpengine socket chosen for this call.
This function can be used from any route.
This function will start playing a media file to one of the endpoints.
Meaning of the parameters is as follows:
flags(string) - a list of flags simialar to the other functions. One of the file, blob or db-id parameters is mandatory to indicate the content of the media file to be played. file is a common choice for specifying rtpengine to get media from a file path, blob to take the content from an inline string and db-id to get the content from the database.
The direction of the media stream is controlled by the from-tag parameter, address (media address from the SDP), or label, if the media stream contains a label. If all of them are missing, the media file is played to the initiator of the SIP request, and will work similar to a ringback tone.
duration_spec(var, optional) - a pseudo variable that will contain the duration of the played file. It will be set to -1 if the duration could not be determined.
sock_var(var, optional) - variable used to store the rtpengine socket chosen for this call.
This function can be used from any route.
Example 1.24. Ringback tone using rtpengine_play_media
...
if (is_method("INVITE") && !has_totag())
rtpengine_play_media("file=/path/to/ringback_tone_file.wav");
...
Example 1.25. Manage music on hold using rtpengine_play_media
...
if (is_method("INVITE") && has_totag()) {
if (is_audio_on_hold()) {
$dlg_val(on_hold) = "1";
rtpengine_play_media("from-tag=$tt file=/path/to/moh_file.wav");
} else if ($dlg_val(on_hold) == "1") {
$dlg_val(on_hold) = "0";
rtpengine_stop_media("from-tag=$tt");
}
}
...
This function will stop playing a media file previously started
by a rtpengine_play_media() call. The meaning
of its parameters is similar to the previous functions. Note that this
function should be called with similar parameters as its matching
rtpengine_play_media() call, otherwise
RTPEngine will not be able to stop media playing.
This function can be used from any route.
Example 1.26. Ringback tone stop using rtpengine_stop_media
...
if (is_method("INVITE") && $rs == 200)
rtpengine_stop_media();
...
This function will block the media sent from one of the endpoints. The direction to be blocked is controled by the flags parameter, the from-tag value.
This function can be used from any route.
This function will resume/unblock the media sent from one of the endpoints. The direction to be blocked is controled by the flags parameter, the from-tag value.
This function can be used from any route.
This function will block the DTMF media sent from one of the endpoints. The direction to be blocked is controled by the flags parameter, the from-tag value.
This function can be used from any route.
This function will resume/unblock the DTMF media sent from one of the endpoints. The direction to be blocked is controled by the flags parameter, the from-tag value.
This function can be used from any route.
This function will start forwarding the media to a TLS destination specified in the tls-send-to parmeter of RTPEngine. This function allows you to select the media stream to forward, by specifing the from-tag of the entity you want to forward the media. If missing, all media streams are forwarded.
This function can be used from any route.
This function will stop forwarding of the media previously started using the rtpengine_start_forwarding() function.
This function can be used from any route.
This function instructs RTP to send the DTMF code to the participant of the call. The code can be a digit (“0-9”) or a special character (one of “*,#,A,B,C,D”). Additional parameters can be configured using the flags parameter. For more information, please consult the RTP documentation.
NOTE: if you are planning to inject DTMF in a session, you have to specify the inject-DTMF flag when the session is created.
This function can be used to convert SIP INFO DTMF keys to RTP DTMF.
This function can be used from any route.
Example 1.33. Example of rtpengine_play_dtmf usage
...
rtpengine_play_dtmf("0"); # send the 0 code upstream
...
Returns the RTP statistics from the RTP proxy. The RTP statistics from the RTP proxy are provided as a string and it does contain several packet counters.
Returnes one of the pre-fined statistics listed below:
MOS-average - without an index, it returns the average MOS value, expressed in an integer between 0 and 50, of all the RTP streams involved in the call, both caller and callee. If index is specified, it has to be one of the from-tag or to-tag involved in the call. In this case, the variable will return the average MOS of all the streams generated by that endpoint with the associated tag value. If you need more granular statistics, check the $rtpquery variable.
jitter-average - similar behavior with MOS-average, but returnes the average jitter.
roundtrip-average - similar behavior with MOS-average, but returnes the average roundtrip.
packetloss-average - similar behavior with MOS-average, but returnes the average packet loss.
MOS-min - without an index, it returns the minimum MOS value (integer value between 0 and 50) of all RTP streams involved in the call, both caller and callee. If the index is specified, it has the same effect as for MOS-average.
jitter-min - similar behavior with MOS-min, but returnes the minimum jitter of a leg/call.
roundtrip-min - similar behavior with MOS-min, but returnes the minimum roundtrip of a leg/call.
packetloss-min - similar behavior with MOS-min, but returnes the minimum packet loss of a leg/call.
MOS-max - without an index, it returns the maximum MOS value (integer value between 0 and 50) of all RTP streams involved in the call, both caller and callee. If the index is specified, it has the same effect as for MOS-average.
jitter-max - similar behavior with MOS-max, but returnes the maximum jitter of a leg/call.
roundtrip-max - similar behavior with MOS-max, but returnes the maximum roundtrip of a leg/call.
packetloss-max - similar behavior with MOS-max, but returnes the maximum packet loss of a leg/call.
MOS-min-at - without an index, it returns the time in seconds elapsed from the start of the call when the MOS value is minimum. If the index is specified, it has the same effect as for MOS-average.
jitter-min-at - similar behavior with MOS-min-at, but returnes the time when the minimum jitter was detected.
roundtrip-min-at - similar behavior with MOS-min-at, but returnes the time when the minimum roundtrip was detected.
packetloss-min-at - similar behavior with MOS-min-at, but returnes the time when the minimum packet loss of a leg/call was detected.
MOS-max-at - without an index, it returns the time in seconds elapsed from the start of the call when the MOS value is maximum. If the index is specified, it has the same effect as for MOS-average.
jitter-max-at - similar behavior with MOS-max-at, but returnes the time when the maximum value of jitter was detected.
roundtrip-max-at - similar behavior with MOS-max-at, but returnes the time when the maximum value of roundtrip was detected.
packetloss-min-at - similar behavior with MOS-max-at, but returnes the time when the maximum packet loss of a leg/call was detected.
NOTE: all these statistics are computed based on the statistics generated by RTPEngine. Some of them might not be available for all the calls (i.e. MOS cannot be computed if the call is too short, or if the phones do not properly report RTP statistics over RTCP). In these cases the variable returns the NULL value.
Example 1.35. $rtpstat(STAT)
...
xlog("Average MOS of the entire call is $rtpstat(MOS-average)\r\n");
xlog("Average MOS of caller is $(rtpstat(MOS-average)[$ft])\r\n");
xlog("Average MOS of callee is $(rtpstat(MOS-average)[$tt])\r\n");
xlog("Min MOS of caller is $(rtpstat(MOS-min)[$ft]) reported at $(rtpstat(MOS-min-at)[$ft])\r\n");
...
Does a Query command to the RTP proxy and returns the answer in a JSON format. You can use this variable to fetch arbitrary data from the RTP proxy such as raw statistics about the call, or other indicators.
You can use a $json() variable to parse its output and extract any information from the query, such as RTP statistics, or MOS values.
Example 1.36. $rtpquery Usage
...
$json(reply) := $rtpquery;
xlog("Total RTP Stats: $json(reply/totals)\n");
...
Enables/disables a RTP proxy.
Parameters:
url - the RTP proxy url (exactly as defined in the config file).
enable - 1 - enable, 0 - disable the RTP proxy.
setid (optional) the set ID of the nodes to be updated. If provided, only nodes in the provided set will be updated.
NOTE: if a RTP proxy is defined multiple times (in the same or different set), all of its instances will be enabled/disabled IF no set ID is provided.
Example 1.37.
rtpengine_enable usage
... ## disable all rtpengines by URL $ opensips-cli -x mi rtpengine_enable udp:192.168.2.133:8081 0 ## enable rtpengine by URL and set ID (3) $ opensips-cli -x mi rtpengine_enable url=udp:192.168.2.133:8081 enable=1 setid=3 ...
Displays all the RTP proxies and their information: set and status (disabled or not, weight and recheck_ticks).
No parameter.
Reloads all rtpengine sets from the database. Used only when the “db_url” parameter is set.
Parameters:
type (optional) soft - when reloading nodes from the database, reuse any existing sockets and keep existing node disabled state. If not provided, then all nodes and sockets will first be torndown and then nodes will be loaded from the database.
No parameter.
Example 1.39.
rtpengine_reload usage
... $ opensips-cli -x mi rtpengine_reload $ opensips-cli -x mi rtpengine_reload type=soft ...
Terminates the SIP dialog by the SIP Call-ID given as parameter.
Parameters:
callid - SIP Call-ID.
Note this is a just a wrapper function over the “dlg_end_dlg” MI function provided by the “dialog” module. This wrapping is done just to make rtpengine happy when trying to terminate SIP calls based on RTP timeouts.
Example 1.40.
teardown usage
... $ opensips-cli -x mi teardown Y2IwYjQ2YmE2ZDg5MWVkNDNkZGIwZjAzNGM1ZDY0ZDQ ...
This event is raised when a notification is received from RTPengine.
Parameters represent the nodes within the Json request received from RTPengine. Common values are:
type - identifies the type of notification (i.e. DTMF)
callid - the callid of the call this event is triggered for
source_tag - from tag of the call this event is triggered for
timestamp - timestamp when the event was triggered
For a DTMF event received, you will also get the following nodes:
source_ip - the IP that triggered the DTMF
event - the event/digit pressed
duration - how long the digit was pressed
volume - volume of the tone
2.1. | How do I migrate from “rtpproxy” or “rtpproxy-ng” to “rtpengine”? |
For the most part, only the names of the functions have changed, with “rtpproxy” in each name replaced with “rtpengine”. For example, “rtpproxy_manage()” has become “rtpengine_manage()”. A few name duplications have also been resolved, for example there is now a single “rtpengine_delete()” instead of “unforce_rtp_proxy()” and the identical “rtpproxy_destroy()”. The largest difference to the old module is how flags are passed to “rtpengine_offer()”, “rtpengine_answer()”, “rtpengine_manage()” and “rtpengine_delete()”. Instead of having a string of single-letter flags, they now take a string of space-separated items, with each item being either a single token (word) or a “key=value” pair. For example, if you had a call “rtpproxy_offer("FRWOC+PS");”, this would then become:
rtpengine_offer("force trust-address symmetric replace-origin replace-session-connection ICE=force RTP/SAVPF");
Finally, if you were using the second parameter (explicit media address) to any of these functions, this has been replaced by the “media-address=...” option within the first string of flags. | |
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) | 222 | 121 | 6034 | 3088 |
| 2. | Bogdan-Andrei Iancu (@bogdan-iancu) | 31 | 17 | 423 | 595 |
| 3. | John Burke (@john08burke) | 25 | 17 | 647 | 102 |
| 4. | Liviu Chircu (@liviuchircu) | 20 | 16 | 91 | 173 |
| 5. | Richard Fuchs | 20 | 2 | 640 | 723 |
| 6. | Vlad Patrascu (@rvlad-patrascu) | 15 | 7 | 218 | 330 |
| 7. | Peter Lemenkov (@lemenkov) | 10 | 7 | 27 | 62 |
| 8. | Eric Tamme (@etamme) | 7 | 5 | 42 | 19 |
| 9. | Nick Altmann (@nikbyte) | 6 | 4 | 43 | 2 |
| 10. | Ovidiu Sas (@ovidiusas) | 5 | 3 | 37 | 7 |
All remaining contributors: Maksym Sobolyev (@sobomax), Zero King (@l2dy), Rob Gagnon (@rgagnon24), Flavio E. Goncalves, Dan Pascu (@danpascu), Oliver Severin Mulelid-Tynes (@olivermt).
(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. | Razvan Crainea (@razvancrainea) | Jun 2014 - Jul 2024 |
| 2. | Liviu Chircu (@liviuchircu) | Jul 2014 - May 2023 |
| 3. | Maksym Sobolyev (@sobomax) | Jan 2021 - Feb 2023 |
| 4. | Peter Lemenkov (@lemenkov) | Jun 2018 - Apr 2022 |
| 5. | John Burke (@john08burke) | Jun 2019 - Apr 2022 |
| 6. | Bogdan-Andrei Iancu (@bogdan-iancu) | Jun 2014 - Apr 2022 |
| 7. | Nick Altmann (@nikbyte) | May 2021 - May 2021 |
| 8. | Flavio E. Goncalves | Oct 2020 - Oct 2020 |
| 9. | Zero King (@l2dy) | Mar 2020 - Sep 2020 |
| 10. | Ovidiu Sas (@ovidiusas) | Jun 2020 - Jun 2020 |
All remaining contributors: Vlad Patrascu (@rvlad-patrascu), Dan Pascu (@danpascu), Oliver Severin Mulelid-Tynes (@olivermt), Rob Gagnon (@rgagnon24), Eric Tamme (@etamme), Richard Fuchs.
(1) including any documentation-related commits, excluding merge commits
Last edited by: Razvan Crainea (@razvancrainea), Liviu Chircu (@liviuchircu), John Burke (@john08burke), Nick Altmann (@nikbyte), Flavio E. Goncalves, Peter Lemenkov (@lemenkov), Vlad Patrascu (@rvlad-patrascu), Bogdan-Andrei Iancu (@bogdan-iancu), Richard Fuchs.
Documentation Copyrights:
Copyright © 2013-2014 Sipwise GmbH
Copyright © 2010 VoIPEmbedded Inc.
Copyright © 2009-2014 TuTPro Inc.
Copyright © 2005 Voice Sistem SRL
Copyright © 2003-2008 Sippy Software, Inc.