Table of Contents
iptrtpproxy_alloc(gate_a_to_b [, existing_sess_ids])
iptrtpproxy_update(gate_a_to_b, session_ids)
iptrtpproxy_adjust_timeout(gate_a_to_b, session_ids)
iptrtpproxy_delete(session_ids)
iptrtpproxy_authorize_media()
iptrtpproxy_set_param(param, value)
iptrtpproxy_set_param("(aggregation/switchboard)_by_sip_ip_(a/b)", sip_ip)
iptrtpproxy_set_param("protected_session_ids", sess_ids)
iptrtpproxy_set_param("o_name", value)
iptrtpproxy_set_param("o_addr", value)
iptrtpproxy_set_param("codec_set", value)
iptrtpproxy_set_param("remove_codec_mask", value)
List of Examples
Table of Contents
iptrtpproxy_alloc(gate_a_to_b [, existing_sess_ids])
iptrtpproxy_update(gate_a_to_b, session_ids)
iptrtpproxy_adjust_timeout(gate_a_to_b, session_ids)
iptrtpproxy_delete(session_ids)
iptrtpproxy_authorize_media()
iptrtpproxy_set_param(param, value)
iptrtpproxy_set_param("(aggregation/switchboard)_by_sip_ip_(a/b)", sip_ip)
iptrtpproxy_set_param("protected_session_ids", sess_ids)
iptrtpproxy_set_param("o_name", value)
iptrtpproxy_set_param("o_addr", value)
iptrtpproxy_set_param("codec_set", value)
iptrtpproxy_set_param("remove_codec_mask", value)
This module provides similar functionality as nathelper but communicates with netfilter kernel xt_RTPPROXY module using the libipt_RTPPROXY userspace library. All RTP streams are manipulated directly in kernel space, no data is copied from kernel to userspace and back, it reduces load and delay. See http://www.2p.cz/en/netfilter_rtp_proxy for more details.
This Kamailio module is written as a light-weight module, there is no dialog management as in Nathelper. The reason is that such an API should be provided by core or a specialized dialog manager module. Because such module is not in git, session information may be stored in extra attributes of the avp_db module and the session id itself in record route as cookie, see the rr module.
It should be able to support all cases as re-invites when SIP client offers media change in SDP and when number of medias in offer/answer are different.
Nathelper may be still used for testing if client is behind the NAT.
There is also support for media authorization. Number of codec sets may be defined. When a message containing SDP offer/answer is being processed then current codecs and streams may be inspected, removed or signallized according a codec set.
Limitations:
Only IPv4 addresses are supported.
More media streams per session supported
The following libraries or applications must be installed before running Kamailio with this module loaded:
netfilter xt_RTPPROXY & libipt_RTPPROXY, see http://www.2p.cz/en/netfilter_rtp_proxy
The modules Makefile must be edited and iptdir setup to the directory with the iptable sources (if different from ~/iptables). Alternatively compile the module using:
make -C modules/iptrtpproxy iptdir=path_to_iptables_src
References iptrtpproxy.cfg, see iptrtpproxy_helper. Default value is /etc/iptrtpproxy.cfg. If only codec authorization is to be used then /dev/null may be used.
References xt_RTPPROXY switchboard for usage by ser module.
The format is:
"name=" value * ( ";" name "=" value ) name = "aggregation" | "sip-addr-"
The name is the switchboard name as declared in config and will be used by script functions and references switchboard. It's mandatory parameter. The special name * set values for all switchboards.
The sip-addr is address used by iptrtpproxy_ser_param(by_sip_ip) to find a switchboard for particular
connection. If not explicitly configured then RTP switchboard gate address are used for this feature.
The aggregation enables to aggregate more switchboards in cluster and to widden bandwidth. Aggregation will take sip-addr from the first switchboard of its.
Example 1.1. Declare switchboard
...
modparam("iptrtpproxy", "config", "/etc/iptrtpproxy.cfg");
modparam("iptrtpproxy", "switchboard", "name=my1;sip-addr-a=1.2.3.4;sip-addr-b=5.6.7.8");
modparam("iptrtpproxy", "switchboard", "name=my2;sip-addr-a=2.3.4.5;sip-addr-b=3.4.5.6;aggregation=my23");
modparam("iptrtpproxy", "switchboard", "name=my3;aggregation=my23");
modparam("iptrtpproxy", "switchboard", "name=*;aggregation=my123");
...
Timeout in seconds used for rerequest remote RTP proxy via RPC command after preceeding error. In other words if a RPC server is unresponsive at the moment then next attempt will be forced after this timeout. Default value is 30.
The hostname used by RPC to identify machine where Ser is running to communicate which RTP proxy via local interface. Default value is taken from system hostname.
There are basic implicit codecs compiled in module, more codecs may be added by this parameter (one codec per modparam).
Declares new codec set. Codecs are declared for each media type independently.
The format is:
"name=" value * ( ";" name "=" value ) name = "media_type" | "rights" | "codecs" | "max_streams" | ( "rtp" | "rtcp" ) "_" ( "bytes" | "packets" ) media_types = "audio" | "video" | "application" | "text" | "message" | "data" | "control" | "?" | "*"
The name is the codec set name to be defined.
The media_type belongs to type at m= SDP line. Question mark means "unknown media" type and asterisk "all media types".
The max_streams defines how many streams (m= lines) is allowed per media type.
The rights defines if particular codec is allowed 0, disallowed, i.e. will be removed
if bit AND operation with remove_codec_mask is non-zero or its presence will be signallized by @iptrtpproxy.auth_rights (any other value).
The codecs comma separated list of codecs. Previous media_type&rights will be applied.
The rtp/rtcp_bytes/packets limits bandwidth per media_type (0 is unlimited). It will override
bandwidth limited by iptrtpproxy_set_param("throttle_*").
Example 1.2. Declare codec_set
...
# enable all codecs, default state when codec is declared
modparam("iptrtpproxy", "codec_set", "name=cs1;media_type=*;max_streams=9999;rights=0;codecs=*");
# allow only 2 audio and 1 video stream
modparam("iptrtpproxy", "codec_set", "name=cs2;media_type=*;max_streams=0;media_type=audio;max_streams=2;media_type=video;max_streams=1");
# dtto, allow only a few audio and video codecs, GSM codec is allowed but signallized
modparam("iptrtpproxy", "codec_set", "name=cs3;media_type=*;max_streams=0;rights=1;codecs=*;media_type=audio;max_streams=2;rights=0;codecs=PCMU,G729,G728,parityfec,telephone-events;rights=2;codecs=GSM;media_type=video;max_streams=1;rights=0;codecs=jpeg,parityfec");
# limit max. bandwidth for video¨
modparam("iptrtpproxy", "codec_set", "name=cs4;media_type=video;rtp_bytes=10000;rtcp_bytes=1000");
...
Parses SDP content and allocates for each RTP media stream one RTP proxy session.
SDP is updates to reflect allocated sessions. Switchboard/aggregation is set using
iptrtpproxy_set_param(by_sip_ip) or iptrtpproxy_set_param("switchboard/aggregation").
Aggregation supports load balancing among more RTP proxies controlled by RPC. The module try to allocate at machines/switchboards in following order (priorities) not yet asked (or being heartbeated) machines, responsive machines, switchboards having percentualy more free slots, non responsive machines.
Proxy may hide caller identity provided at o= line using
@iptrtpproxy.o_name/addr and iptrtpproxy_set_param(o_name/addr)
functions. But the script is responsible for rewritting to original values in
a response or a callee initiated re-INVITE. Therefore original value need to be stored
in-dialog.
if gate_a_to_b bit 0 is set then SDP regards to gate-a to gate-b direction.
protected_session_ids list of existing sessions enables reusing already allocated sessions in re-INVITE without allocating new sessions for each stream in SDP regardless a IP/port is required. It's mostly undesirable, typically "hold-on" is done via re-INVITE without any change. There is drawback because callee cannot change IP:port in 200OK which is legal case in RFC3264. But because some non-RFC3264 compliant phones dislike proactively changed IP:port at RTP proxy it seems it's less evil.
function returns true is a session was created, identifier is available
via select @iptrtpproxy.session_ids.
Example 1.3. iptrtpproxy_alloc usage
...
if (!iptrtpproxy_set_param("aggregation_by_sip_ip_a", "@received.ip")) {
if (!iptrtpproxy_set_param("switchboard_by_sip_ip_a", "@received.ip")) {
t_reply("500", "RTP proxy error");
drop;
}
}
eval_push("x:%@next_hop.src_ip");
if (@eval.get[-1] == @received.ip) {
if (@iptrtpproxy.aggregation_a) {
iptrtpproxy_set_param("aggregation_b", "@iptrtpproxy.aggregation_a");
}
else {
iptrtpproxy_set_param("switchboard_b", "@iptrtpproxy.switchboard_a");
}
}
else {
if (!iptrtpproxy_set_param("aggregation_by_sip_ip_b", "@eval.get[-1]")) {
if (!iptrtpproxy_set_param("switchboard_by_sip_ip_b", "@eval.get[-1]")) {
t_reply("500", "RTP proxy error");
drop;
}
}
}
eval_remove(-1, 1);
if (!iptrtpproxy_alloc("1")) {
t_reply("500", "RTP proxy error");
drop;
}
$sess_ids = @iptrtpproxy.session_ids;
...
Parses SDP content and updates sessions provided by session_ids and
updates SDP. If succesfull then session_ids may be changed (in case e.g. media
stream has port zero particular session is released), the
result of @iptrtpproxy.session_ids should be stored for future in-dialog usage.
The SDP contect is also affected by iptrtpproxy_set_param(o_name/addr)
functions. If a stream is deactivated in SDP then Sessions may be deleted unless
mentioned in protected_session_ids.
if gate_a_to_b bit 0 is set then SDP regards to gate-a to gate-b direction.
if gate_a_to_b bit 1 is set then SDP is updated only, no RTP session are affected. Should be used when handling retransmission in onreply route, retransmission replies are not eaten be tm module!
Example 1.4. iptrtpproxy_update usage
...
# load $sess_ids from dialog
if (iptrtpproxy_update("0", $sess_ids)) {
$sess_ids = @iptrtpproxy.session_ids;
# save sess_ids in dialog
}
...
Adjust timeout for particular gate. It's useful in "200 OK" decrease timeout to learning timeout if INVITE has set (long) ringing timeout.
if gate_a_to_b bit 0 is set then it regards to gate-a to gate-b direction.
Example 1.5. iptrtpproxy_adjust_timeout usage
...
# load $sess_ids from dialog
if (status=~"18[0-9]") {
iptrtpproxy_set_param("learning_timeout", "60");
}
else {
iptrtpproxy_set_param("learning_timeout", "5");
}
if (iptrtpproxy_adjust_timeout("0", $sess_ids)) {
}
...
Delete sessions identified by session_ids. May be used when dialog is being destroyed (BYE) or when INVITE failed in failure route. If protected_session_ids list is provided then this set is excluded from sessions to be deleted.
Example 1.6. iptrtpproxy_delete usage
... # load $sess_ids from dialog iptrtpproxy_delete($sess_ids); ...
Authorizes SDP media according currect codec_set. If bit AND operation
between rights in codec set and remove_codec_mask is non zero then
such a codec are to be removed. The result may be obtained from
@iptrtpproxy.auth_rights which returns max. right which has been applied when
processing all codecs of enabled streams.
The function MUST NOT be called after iptrtpproxy_alloc/update!
But the function may be called several times to authorize using more codec sets.
Example 1.7. iptrtpproxy_authorize_media usage
...
if (@iptrtpproxy.active_media_num == "0") break;
iptrtpproxy_set_param("codec_set", "cs2");
iptrtpproxy_set_param("remove_codec_mask", "1");
if (!iptrtpproxy_authorize_media()) {
t_reply("415", "Cannot authorize media");
drop;
}
iptrtpproxy_set_param("codec_set", "cs3");
if (!iptrtpproxy_authorize_media()) {
t_reply("415", "Cannot authorize media");
drop;
}
if (@iptrtpproxy.active_media_num == "0") {
t_reply("488", "Not acceptable here");
drop;
}
if (@iptrtpproxy.auth_rights == "2") {
append_hf_value("Contact: <sip:1.2.3.4>");
t_reply("301", "Redirect to transcoder");
drop;
}
...
Set particular parameter needed mainly by iptrtpproxy_alloc/update/adjust_timeout.
The paramter value is availble via @iptrtpproxy.<param>.
Supported parameters: expiration_timeout, ttl, learning_timeout, always_learn, aggregation_a, aggregation_b, switchboard_a, switchboard_b, throttle_mark, throttle_rtp_max_bytes, throttle_rtp_max_packets, throttle_rtcp_max_bytes, throttle_rtcp_max_packets.
Find corresponding aggregation or switchboard and set
@iptrtpproxy.aggregation_a/b or @iptrtpproxy.switchboard_a/b.
Switchboards/aggregations are compared against sip-addr,
it allow separate SIP and RTP traffic and RTP aggregation.
sip_ip IP to be compared, typically @received.ip
or @next_hop.src_ip.
function returns true if switchboard/aggregation was found
Used for reusing sessions in iptrtpproxy_alloc, iptrtpproxy_update
and iptrtpproxy_delete.
Username to be rewritten at o= line by iptrtpproxy_alloc/update
to hide caller identity. If value is blank then username is left unchanged.
Address to be rewritten at o= line by iptrtpproxy_alloc/update
to hide caller identity. If value is blank then address is left unchanged.
Codec set for iptrtpproxy_authorize_media. Current codec set
may be obtained by @iptrtpproxy.codec_set.
Returns sessions allocated/updated in iptrtpproxy_alloc/update.
The format is:
switchboard_name [ ":" [ session_id "/" created ] * ( "," session_id "/" created ) ] ] session_id = * ( [0-9] ) ; empty when no session allocated created = timestamp