rtpproxy man page on DragonFly
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RTPPROXY(8) [FIXME: manual] RTPPROXY(8)
NAME
rtpproxy - RTP (Real-time Transport Protocol) Proxy Server
SYNOPSIS
rtpproxy [-?] [-2] [-f] [-v] [-V] [-R] [-l addr1[/addr2]]
[-6 addr1[/addr2]] [-s ctrl_socket] [-t tos] [-p pidfile]
[-T max_ttl] [-r rdir [-S sdir]] [-L nofile_limit]
[-A advaddr1[/advaddr2]] [-m min_port] [-M max_port]
[-u uname[:gname]] [-w sock_mode] [-F] [-i]
[-n timeout_socket] [-P] [-a] [-d log_level[:log_facility]]
[-W setup_ttl]
DESCRIPTION
rtpproxy is a symmetric RTP proxy designed to be used in conjunction
with the SIP Express Router (SER) or any other SIP proxy or SIP B2BUA
capable of rewriting SDP bodies in SIP messages that it processes.
The main purpose of rtpproxy is to make the communication between SIP
user agents behind NAT(s) (Network Address Translator) possible.
Several cases exists when direct end-to-end communication is not
possible and RTP streams have to be relayed through another host.
Rtpproxy can be used to setup such a relaying host.
When two listen interfaces have been specified using the command line
parameters described below then rtpproxy will enter so called bridging
mode. In briding mode rtpproxy forwards RTP packets received on one
interface to the other interface and vice versa. This mode can be used
to forward RTP packets between networks without direct network level
connectivy (provided that the host running rtpproxy has one interface
in both of them). One particular application of bridging mode is
IPv4/IPv6 traversal of RTP packets.
When instructured by SER rtpproxy can also record the entire RTP
session in a file on a local harddisk or play a pre-recorded file to
the user agent (so called Music-on-Hold).
OPTIONS
-?
Show summary of options.
-2
Send every RTP packet twice in sessions that use low-bitrate
codecs. Only packets that are smaller than 128 bytes will be sent
twice. This option can improve audio quality on lossy links.
-f
Rtpproxy will stay in foreground mode if this option is set.
-v
Show version of program.
-V
Show command protocol version.
-l addr1[/addr2]
IPv4 listen IP address(es). You can specify either one or two
addresses. If two addresses have been specified then rtpproxy will
work in bridging mode.
-6 addr1[/addr2]
IPv6 listen IP address(es). You can specify either one or two
addresses. If two addresses have been specified then rtpproxy will
work in bridging mode.
-s ctrl_socket
This parameter configures rtpproxy control socket. The control
socket is used by nathelper module of SER to create/modify/delete
RTP sessions to be relayed. Format of ctrl_socket is
<type>:<socket>. Following types are supported:
· udp: Create UDP control socket. In this mode RTPProxy will
listen on UDP for control messages from SER/nathelper.
Example: -s udp:127.0.0.1:9000
IP address can be '*' in which case rtpproxy will listen on all
local interfaces. If omitted port 22222 is used.
Note
RTPProxy control protocol has no built-in security
mechanisms. Make sure that you protect the listening IP and
port properly when using RTPProxy with UDP control socket.
· udp6: Create IPv6 UDP control socket. In this mode RTPProxy
will listen on UDP/IPv6 for control messages from
SER/nathelper.
Example: -s udp6:::1:9000
· unix: Create UNIX domain socket for control interface. In this
mode SER/nathelper and RTPProxy must be running on the same
host. This is the default setting for both SER/nathelper and
rtpproxy.
Example: -s unix:/var/run/rtpproxy.sock
Default value is /var/run/rtpproxy.sock.
-t tos
Set ToS (Type of Service) in the outgoing UDP packets to this
value. Default value is 0xB8. Setting this parameter to -1 disables
setting ToS resulting in operating system default ToS being used
instead.
-r rec_dir
Directory where recorded RTP sessions will be stored.
-S spool_dir
Spool directory for RTP sessions being recorded. The file will be
moved to directory configured in -r option after the session
finishes.
-R
Do not record RTCP when recording an RTP session. This option is
disabled (rtpproxy will record RTCP) by default.
-p pid_file
This parameter configures the name of the file where PID of running
rtpproxy will be stored. Default is /var/run/rtpproxy.pid.
-T max_ttl
Limit the maximum TTL (Time To Live) of outgoing IP packets to the
value of max_ttl.
-L nofile_limit
Adjust the number of simultaneous open connections. Please note
that each RTP media stream requires four open connections. A SIP
call can open more than one RTP media stream depending on the
client's setup.
-A advaddr1[/advaddr2]
Setup advertised address if necessary.
-m min_port
Set lower limit on UDP ports range that the RTPproxy uses for
RTP/RTCP sessions to min_port. Default is 35000.
-M max_port
Set upper limit on UDP ports range that the RTPproxy uses for
RTP/RTCP sessions to max_port. Default is 65000.
-u uname[:gname]
Switch RTPproxy to UID identified by the uname and optional GID
identified by gname when proxy is up and running.
-w sock_mode
Set access mode for the controlling UNIX-socket (if used). Only
applies if RTPproxy runs under a different GID using -u option.
-F
By default the RTPproxy will warn user if running as superuser (UID
0) in local control mode and refuse to run in remote control mode
at all. This switch removes the check.
-i
Enable independent timeout mode. By default, a timeout (which
results in automatic destruction of the session) can only occur if
no RTP packets are received on any of the session's ports. This
option if set varies that behaviour, such that a timeout will occur
if packets are still being received on one port but not the other.
The option should be used with caution since in some cases it's
perfectly fine to have packets coming from only one side of
conversation (i.e. when the second party has muted its audio).
-n timeout_socket
This parameter configures the optional timeout notification socket.
The socket should be created by another application, preferably
before starting rtpproxy. For those sessions where the timeout
mechanism is enabled, notifications are sent on this socket if the
session times out.
Format of timeout_socket is <type>:<socket>. Following types are
supported:
· unix:Connect to UNIX domain socket for sending timeout
notifications. In this mode B2BUA and RTPProxy must be running
on the same host.
Example: -n unix:/var/run/rtpproxy_timeout.sock
· tcp:Connect to a remote host using TCP/IP for sending timeout
notifications. Format of the socket parameter in this case is
<host>:<port>.
Example: -n tcp:10.20.30:12345
There is no default value, notifications are not sent and not
permitted unless a value is specified explicitly.
-P
Record sessions using PCAP file format instead of non-standard
ad-hoc format. The PCAP format, which is the de-facto standard for
packet capturing software, has the advantage of being compatible
with numerous third-party tools and utilities, such as Wireshark
(Ethereal) for example. The drawback of PCAP is sligtly larger
overhead (extra 12 bytes for every saved RTP packet for IPv4).
Also, recording IPv6 sessions in PCAP format is not supported at
the moment.
-a
Record all sessions going through the RTPproxy unconditionally. By
default the RTPproxy requires call control software (i.e. SER,
OpenSER or B2BUA) to enable recording explicitly on per-session
basis by sending appropriate record command.
-d log_level[:log_facility]
This parameter configures the verbosity level of the log output.
Possible log_level values in the order from the most verboe to the
least verbose are: DBUG, INFO, WARN, ERR and CRIT.
The optional log_facility parameter sets syslog(3) facility
assigned to log messages.
Example: -d WARN:LOG_LOCAL5
The default level in foreground mode is is DBUG, in background -
WARN and facility is LOG_DAEMON.
HOWITWORKS
When SER receives an INVITE request, it extracts Call-ID from it and
communicates it to rtpproxy via Unix domain socket or UDP. Rtproxy
looks for an existing session with such Call-ID. If the session exists
it returns UDP port for that session, if not, then it creates a new
session, binds to a first empty UDP port from the range specified at
the compile time and returns number of that port to a SER. After
receiving reply from the proxy, SER replaces media ip:port in the SDP
to point to the proxy and forwards request as usually.
When SER receives a non-negative SIP reply with SDP it again extracts
Call-ID from it and communicates it to the proxy. In this case the
proxy does not allocate a new session if it doesn't exist, but simply
performs a lookup among existing sessions and returns either a port
number if the session is found, or error code indicating that there is
no session with such id. After receiving positive reply from the proxy,
SER replaces media ip:port in the SIP reply to point to the proxy and
forwards reply as usually.
After the session has been created, the proxy listens on the port it
has allocated for that session and waits for receiving at least one UDP
packet from each of two parties participating in the call. Once such
packet is received, the proxy fills one of two ip:port structures
associated with each call with source ip:port of that packet. When both
structures are filled in, the proxy starts relaying UDP packets between
parties.
The proxy tracks idle time for each of existing sessions (i.e. the time
within which there were no packets relayed), and automatically cleans
up a sessions whose idle times exceed the value specified at compile
time (60 seconds by default).
FILES
/usr/sbin/rtpproxy
LICENSE
This program is licensed under the BSD license. See COPYING file in the
rtpproxy sources for details.
AVAILABILITY
The latest version of this program can be found at
http://www.rtpproxy.org/.
SEEALSO
ser(8).
AUTHOR
Maxim Sobolev
Author.
COPYRIGHT
Copyright © 2006 janakj
[FIXME: source] Jun 16, 2008 RTPPROXY(8)
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