Copyright © 2003-2008 Sippy Software, Inc.
Copyright © 2005 Voice Sistem SRL
Copyright © 2009-2012 TuTPro Inc.
Copyright © 2010 VoIPEmbedded Inc.
Copyright © 2013 Sipwise GmbH
Table of Contents
List of Examples
rtpproxy_sock
parameterrtpproxy_disable_tout
parameterrtpproxy_tout
parameterrtpproxy_retr
parameterextra_id_pv
parameterset_rtp_proxy_set
usagertpproxy_offer
usagertpproxy_answer
usagertpproxy_destroy
usagertpproxy_manage
usagestart_recording
usagenh_enable_rtpp
usagenh_show_rtpp
usageTable of Contents
This 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 ngcp-rtpproxy-ng https://github.com/sipwise/mediaproxy-ng. The rtpproxy-ng 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 rtpproxy-ng module can support multiple RTP proxies for balancing/distribution and control/selection purposes.
The module allows definition of several sets of rtpproxies. 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 “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 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()!!
The following modules must be loaded before this module:
tm module - (optional) if you want to have rtpproxy_manage() fully functional
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 modparam("rtpproxy-ng", "rtpproxy_sock", "udp:localhost:12221") # multiple rtproxies for LB modparam("rtpproxy-ng", "rtpproxy_sock", "udp:localhost:12221 udp:localhost:12222") # multiple sets of multiple rtproxies modparam("rtpproxy-ng", "rtpproxy_sock", "1 == udp:localhost:12221 udp:localhost:12222") modparam("rtpproxy-ng", "rtpproxy_sock", "2 == udp:localhost:12225") ...
Once an RTP proxy was found unreachable and marked as disabled, the rtpproxy-ng module will not attempt to establish communication to that RTP proxy for rtpproxy_disable_tout seconds.
Default value is “60”.
Example 1.2. Set rtpproxy_disable_tout
parameter
... modparam("rtpproxy-ng", "rtpproxy_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”.
The parameter sets the PV defination to use when the “b” parameter is used on unforce_rtp_proxy(), rtpproxy_offer(), rtpproxy_answer() or rtpproxy_manage() command.
Default is empty, the “b” parameter may not be used then.
Example 1.5. Set extra_id_pv
parameter
... modparam("rtpproxy-ng", "extra_id_pv", "$avp(extra_id)") ...
Sets the Id of the rtpproxy set to be used for the next unforce_rtp_proxy(), rtpproxy_offer(), rtpproxy_answer() or rtpproxy_manage() command. The parameter can be an integer or a config variable holding 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 - flags to turn on some features.
1 - append first Via branch to Call-ID when sending command to rtpproxy. This can be used to create one media session per branch on the rtpproxy. When sending a subsequent “delete” command to the rtpproxy, you can then stop just the session for a specific branch when passing the flag '1' or '2' in the “unforce_rtpproxy”, 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 rtpproxy 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 ngcp-mediaproxy-ng rtpproxy at the moment!
2 - append second Via branch to Call-ID when sending command to rtpproxy. See flag '1' for its meaning.
3 - behave like flag 1 is set for a request and like flag 2 is set for a reply.
a - flags that UA from which message is received doesn't support symmetric RTP. (automatically sets the 'r' flag)
b - append branch specific variable to Call-ID when sending command to rtpproxy. This creates one rtpproxy session per unique variable. Works similar to the 1, 2 and 3 parameter, but is usefull when forking to multiple destinations on different address families or network segments, requiring different rtpproxy parameters. The variable value is taken from the “extra_id_pv”. When used, it must be used in every call to rtpproxy_manage(), rtpproxy_offer(), rtpproxy_answer() and rtpproxy_destroy() with the same contents of the PV. The b parameter may not be used in conjunction with the 1, 2 or 3 parameter to use the Via branch in the Call-ID.
l - force “lookup”, 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.
i, e - these flags specify the direction of the SIP message. These flags only make sense when rtpproxy is running in bridge mode. 'i' means internal network (LAN), 'e' means external network (WAN). 'i' corresponds to rtpproxy's first interface, 'e' corresponds to rtpproxy's second interface. You always have to specify two flags to define the incoming network and the outgoing network. For example, 'ie' should be used for SIP message received from the local interface and sent out on the external interface, and 'ei' vice versa. Other options are 'ii' and 'ee'. So, for example if a SIP requests is processed with 'ie' flags, the corresponding response must be processed with 'ie' flags.
For ngcp-mediaproxy-ng, these flags are used to select between IPv4 and IPv6 addresses, corresponding to 'i' and 'e' respectively. For example, if the request is coming from an IPv4 host and is going to an IPv6 host, the flags should be specified as 'ie'.
Note: As rtpproxy in bridge mode s per default asymmetric, you have to specify the 'w' flag for clients behind NAT! See also above notes!
x - this flag an alternative to the 'ie' or 'ei'-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 ngcp-mediaproxy-ng.
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.
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 a 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.
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.
+ - instructs the RTP proxy to 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. Without this flag, 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.
- - instructs the RTP proxy to discard any ICE attributes and not insert any new ones into the SDP. Mutually exclusive with the '+' flag.
s, S, p, P - 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 "S" flag indicates that SRTP should be used, while "s" indicates that SRTP should not be used. "P" indicates that the advanced RTCP profile with feedback messages should be used, and "p" indicates that the regular RTCP profile should be used. As such, the combinations "sp", "sP", "Sp" and "SP" select between RTP/AVP, RTP/AVPF, RTP/SAVP and RTP/SAVPF, respectively.
ip_address - new SDP IP address.
This function can be used from ANY_ROUTE.
Example 1.7. 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_answer() 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 ANY_ROUTE.
Meaning of the parameters is as follows:
flags - flags to turn on some features.
1 - append first Via branch to Call-ID when sending command to rtpproxy. This can be used to create one media session per branch on the rtpproxy. When sending a subsequent “delete” command to the rtpproxy, you can then stop just the session for a specific branch when passing the flag '1' or '2' in the “unforce_rtpproxy”, 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 rtpproxy gets an “update” command for a new branch, and then a “delete” command for the previous branch, which would otherwise delete the full call, breaking the subsequent “lookup” for the new branch. This flag is only supported by the ngcp-mediaproxy-ng rtpproxy at the moment!
2 - append second Via branch to Call-ID when sending command to rtpproxy. See flag '1' for its meaning.
b - append branch specific variable to Call-ID when sending
command to rtpproxy. See rtpproxy_offer() for details.
<listitem>
Manage the RTPProxy session - it combines the functionality of rtpproxy_offer(), rtpproxy_answer() and unforce_rtpproxy(), detecting internally based on message type and method which one to execute.
It can take the same parameters as rtpproxy_offer().
The flags parameter to rtpproxy_manage() can be a configuration variable
containing the flags as a string.
Functionality:
If INVITE with SDP, then do rtpproxy_offer()
If INVITE with SDP, when the tm module is loaded, mark transaction with
internal flag FL_SDP_BODY to know that the 1xx and 2xx are for
rtpproxy_answer()
If ACK with SDP, then do rtpproxy_answer()
If BYE or CANCEL, or called within a FAILURE_ROUTE[], then do unforce_rtpproxy()
If reply to INVITE with code >= 300 do unforce_rtpproxy()
If reply with SDP to INVITE having code 1xx and 2xx, then
do rtpproxy_answer()
if the request had SDP or tm is not loaded,
otherwise do rtpproxy_offer()
This function can be used from ANY_ROUTE.
This function will send a signal to the RTP Proxy to record the RTP stream on the RTP Proxy. This function is not supported by ngcp-mediaproxy-ng at the moment!
This function can be used from REQUEST_ROUTE and ONREPLY_ROUTE.
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. The statistics
must be retrieved before the session is deleted (before unforce_rtpproxy()
).
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 of its instances will be enables/disabled.
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. |
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2.2. |
Where can I find more about Kamailio? |
Take a look at http://www.kamailio.org/. |
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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 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? |
Please follow the guidelines provided at: http://sip-router.org/tracker. |