<?xml version="1.0"?>
<feed xmlns="http://www.w3.org/2005/Atom" xml:lang="en">
	<id>https://lms.onnocenter.or.id/wiki/index.php?action=history&amp;feed=atom&amp;title=VoIP_Cookbook%3A_SIPp_COMMANDS</id>
	<title>VoIP Cookbook: SIPp COMMANDS - Revision history</title>
	<link rel="self" type="application/atom+xml" href="https://lms.onnocenter.or.id/wiki/index.php?action=history&amp;feed=atom&amp;title=VoIP_Cookbook%3A_SIPp_COMMANDS"/>
	<link rel="alternate" type="text/html" href="https://lms.onnocenter.or.id/wiki/index.php?title=VoIP_Cookbook:_SIPp_COMMANDS&amp;action=history"/>
	<updated>2026-04-20T02:04:11Z</updated>
	<subtitle>Revision history for this page on the wiki</subtitle>
	<generator>MediaWiki 1.45.1</generator>
	<entry>
		<id>https://lms.onnocenter.or.id/wiki/index.php?title=VoIP_Cookbook:_SIPp_COMMANDS&amp;diff=17357&amp;oldid=prev</id>
		<title>Onnowpurbo: New page: Usage:   sipp remote_host[:remote_port] [options]  Available options:   -v		: Display version and copyright information.  -aa		: Enable automatic 200 OK answer for INFO, UPDATE and NOTIFY ...</title>
		<link rel="alternate" type="text/html" href="https://lms.onnocenter.or.id/wiki/index.php?title=VoIP_Cookbook:_SIPp_COMMANDS&amp;diff=17357&amp;oldid=prev"/>
		<updated>2010-03-09T08:02:12Z</updated>

		<summary type="html">&lt;p&gt;New page: Usage:   sipp remote_host[:remote_port] [options]  Available options:   -v		: Display version and copyright information.  -aa		: Enable automatic 200 OK answer for INFO, UPDATE and NOTIFY ...&lt;/p&gt;
&lt;p&gt;&lt;b&gt;New page&lt;/b&gt;&lt;/p&gt;&lt;div&gt;Usage:&lt;br /&gt;
&lt;br /&gt;
 sipp remote_host[:remote_port] [options]&lt;br /&gt;
&lt;br /&gt;
Available options:&lt;br /&gt;
&lt;br /&gt;
 -v		: Display version and copyright information.&lt;br /&gt;
 -aa		: Enable automatic 200 OK answer for INFO, UPDATE and NOTIFY messages.&lt;br /&gt;
 -auth_uri	: Force the value of the URI for authentication.&lt;br /&gt;
        	  By default, the URI is composed of remote_ip:remote_port.&lt;br /&gt;
 -base_cseq	: Start value of [cseq] for each call.&lt;br /&gt;
 -bg		: Launch SIPp in background mode.&lt;br /&gt;
 -bind_local	: Bind socket to local IP address, i.e. the local IP address is used as the&lt;br /&gt;
                  source IP address. If SIPp runs in server mode it will only listen on the&lt;br /&gt;
                  local IP address instead of all IP addresses.&lt;br /&gt;
 -buff_size	: Set the send and receive buffer size.&lt;br /&gt;
 -calldebug_file : Set the name of the call debug file.&lt;br /&gt;
 -calldebug_overwrite: Overwrite the call debug file (default true).&lt;br /&gt;
 -cid_str	: Call ID string (default %u-%p@%s).  %u=call_number,&lt;br /&gt;
        	  %s=ip_address, %p=process_number, %%=% (in any order).&lt;br /&gt;
 -ci		: Set the local control IP address&lt;br /&gt;
 -cp		: Set the local control port number. Default is 8888.&lt;br /&gt;
 -d		: Controls the length of calls. More precisely, this controls the duration&lt;br /&gt;
                  of &amp;#039;pause&amp;#039; instructions in the scenario, if they do not have a&lt;br /&gt;
                  &amp;#039;milliseconds&amp;#039; section. Default value is 0 and default unit is milliseconds.&lt;br /&gt;
 -deadcall_wait : How long the Call-ID and final status of calls should be kept to improve message&lt;br /&gt;
 		  and error logs (default unit is ms).&lt;br /&gt;
 -default_behaviors: Set the default behaviors that SIPp will use.  Possbile values are:&lt;br /&gt;
 			- all	Use all default behaviors&lt;br /&gt;
 			- none	Use no default behaviors&lt;br /&gt;
 			- bye	Send byes for aborted calls&lt;br /&gt;
 			- abortunexp	Abort calls on unexpected messages&lt;br /&gt;
 			- pingreply	Reply to ping requests&lt;br /&gt;
 		  If a behavior is prefaced with a -, then it is turnedoff.  Example: all,-bye&lt;br /&gt;
 -error_file	: Set the name of the error log file.&lt;br /&gt;
 -error_overwrite : Overwrite the error log file (default true).&lt;br /&gt;
 -f		: Set the statistics report frequency on screen. Default is 1 and default unit is seconds.&lt;br /&gt;
 -fd		: Set the statistics dump log report frequency. Default is 60 and default unit is seconds.&lt;br /&gt;
 -i		: Set the local IP address for &amp;#039;Contact:&amp;#039;,&amp;#039;Via:&amp;#039;, and &amp;#039;From:&amp;#039; headers.&lt;br /&gt;
 	  	  Default is primary host IP address.&lt;br /&gt;
 -inf		: Inject values from an external CSV file during calls into the scenarios.&lt;br /&gt;
 	  	  First line of this file say whether the data is to be read in sequence (SEQUENTIAL),&lt;br /&gt;
 	  	  random (RANDOM), or user (USER) order.&lt;br /&gt;
 	  	  Each line corresponds to one call and has one or more &amp;#039;;&amp;#039; delimited data fields.&lt;br /&gt;
 	  	  Those fields can be referred as [field0], [field1], ... in the xml scenario file. &lt;br /&gt;
 	   	  Several CSV files can be used simultaneously (syntax: -inf f1.csv -inf f2.csv ...)&lt;br /&gt;
 -infindex 	: file field&lt;br /&gt;
 	  	  Create an index of file using field.  For example -inf users.csv -infindex users.csv 0&lt;br /&gt;
 	  	  creates an index on the first key.&lt;br /&gt;
 -ip_field 	: Set which field from the injection file contains the IP address from which the client&lt;br /&gt;
 	  	  will send its messages. If this option is omitted and the &amp;#039;-t ui&amp;#039; option is present,&lt;br /&gt;
 	  	  then field 0 is assumed. Use this option together with &amp;#039;-t ui&amp;#039;&lt;br /&gt;
 -l		: Set the maximum number of simultaneous calls. Once this limit is reached,&lt;br /&gt;
 	  	  traffic is decreased until the number of open calls goes down.&lt;br /&gt;
 	  	  Default: (3 * call_duration (s) * rate).&lt;br /&gt;
 -log_file 	: Set the name of the log actions log file.&lt;br /&gt;
 -log_overwrite : Overwrite the log actions log file (default true).&lt;br /&gt;
 -lost		: Set the number of packets to lose by default&lt;br /&gt;
 		  (scenario specifications override this value).&lt;br /&gt;
 -rtcheck 	: Select the retransmisison detection method: full (default) or loose.&lt;br /&gt;
 -m		: Stop the test and exit when &amp;#039;calls&amp;#039; calls are processed&lt;br /&gt;
 -mi		: Set the local media IP address (default: local primary host IP address)&lt;br /&gt;
 -master 	: 3pcc extended mode: indicates the master number&lt;br /&gt;
 -max_recv_loops  : Set the maximum number of messages received read per cycle.&lt;br /&gt;
        	  Increase this value for high traffic level.  The default value is 1000.&lt;br /&gt;
 -max_sched_loops : Set the maximum number of calsl run per event loop.&lt;br /&gt;
	  	  Increase this value for high traffic level.  The default value is 1000.&lt;br /&gt;
 -max_reconnect  : Set the the maximum number of reconnection.&lt;br /&gt;
 -max_retrans	: Maximum number of UDP retransmissions before call ends on timeout. &lt;br /&gt;
 	  	  Default is 5 for INVITE transactions and 7 for others.&lt;br /&gt;
 -max_invite_retrans: Maximum number of UDP retransmissions for invite transactions&lt;br /&gt;
 	  	  before call ends on timeout.&lt;br /&gt;
 -max_non_invite_retrans: Maximum number of UDP retransmissions for non-invite transactions&lt;br /&gt;
        	  before call ends on timeout.&lt;br /&gt;
 -max_log_size : What is the limit for error and message log file sizes.&lt;br /&gt;
 -max_socket	: Set the max number of sockets to open simultaneously. This option is significant&lt;br /&gt;
 	  	  if you use one socket per call. Once this limit is reached,&lt;br /&gt;
 		  traffic is distributed over the sockets already opened. Default value is 50000&lt;br /&gt;
 -mb		: Set the RTP echo buffer size (default: 2048).&lt;br /&gt;
 -message_file : Set the name of the message log file.&lt;br /&gt;
 -message_overwrite: Overwrite the message log file (default true).&lt;br /&gt;
 -mp		: Set the local RTP echo port number. Default is 6000.&lt;br /&gt;
 -nd		: No Default. Disable all default behavior of SIPp which are the following:&lt;br /&gt;
                  On UDP retransmission timeout, abort the call by sending a BYE or a CANCEL&lt;br /&gt;
                  On receive timeout with no ontimeout attribute, abort the call by sending a BYE or a CANCEL&lt;br /&gt;
                  On unexpected BYE send a 200 OK and close the call&lt;br /&gt;
                  On unexpected CANCEL send a 200 OK and close the call&lt;br /&gt;
                  On unexpected PING send a 200 OK and continue the call&lt;br /&gt;
                  On any other unexpected message, abort the call by sending a BYE or a CANCEL&lt;br /&gt;
 -nr		: Disable retransmission in UDP mode.&lt;br /&gt;
 -nostdin	: Disable stdin.&lt;br /&gt;
 -p		: Set the local port number.  Default is a random free port chosen by the system.&lt;br /&gt;
 -pause_msg_ign : Ignore the messages received during a pause defined in the scenario &lt;br /&gt;
 -periodic_rtd	: Reset response time partition counters each logging interval.&lt;br /&gt;
 -plugin	: Load a plugin.&lt;br /&gt;
 -r		: Set the call rate (in calls per seconds).  This value can bechanged during test&lt;br /&gt;
 		  by pressing &amp;#039;+&amp;#039;,&amp;#039;_&amp;#039;,&amp;#039;*&amp;#039; or &amp;#039;/&amp;#039;. Default is 10.&lt;br /&gt;
 		  pressing &amp;#039;+&amp;#039; key to increase call rate by 1 * rate_scale,&lt;br /&gt;
 		  pressing &amp;#039;-&amp;#039; key to decrease call rate by 1 * rate_scale,&lt;br /&gt;
 		  pressing &amp;#039;*&amp;#039; key to increase call rate by 10 * rate_scale,&lt;br /&gt;
 		  pressing &amp;#039;/&amp;#039; key to decrease call rate by 10 * rate_scale.&lt;br /&gt;
 		  If the -rp option is used, the call rate is calculated with the period in ms&lt;br /&gt;
 		  given by the user.&lt;br /&gt;
 -rp		: Specify the rate period for the call rate.  Default is 1 second and default unit&lt;br /&gt;
 		  is milliseconds.  This allows you to have n calls every m milliseconds&lt;br /&gt;
 		  (by using -r n -rp m).&lt;br /&gt;
 		  Example:	-r 7 -rp 2000 ==&amp;gt; 7 calls every 2 seconds.&lt;br /&gt;
 				-r 10 -rp 5s =&amp;gt; 10 calls every 5 seconds.&lt;br /&gt;
 -rate_scale	: Control the units for the &amp;#039;+&amp;#039;, &amp;#039;-&amp;#039;, &amp;#039;*&amp;#039;, and &amp;#039;/&amp;#039; keys.&lt;br /&gt;
 -rate_increase : Specify the rate increase every -fd units (default is seconds). &lt;br /&gt;
 		  This allows you to increase the load for each independent logging period.&lt;br /&gt;
 		  Example: -rate_increase 10 -fd 10s ==&amp;gt; increase calls by 10 every 10 seconds.&lt;br /&gt;
 -rate_max	: If -rate_increase is set, then quit after the rate reaches this value.&lt;br /&gt;
 		  Example: -rate_increase 10 -rate_max 100 ==&amp;gt; increase calls by 10 until 100 cps is hit.&lt;br /&gt;
 -no_rate_quit	 : If -rate_increase is set, do not quit after the rate reaches -rate_max.&lt;br /&gt;
 -recv_timeout : Global receive timeout. Default unit is milliseconds. If the expected message is not&lt;br /&gt;
 		  received, the call times out and is aborted.&lt;br /&gt;
 -send_timeout : Global send timeout. Default unit is milliseconds. If a message is not sent&lt;br /&gt;
 		  (due to congestion), the call times out and is aborted.&lt;br /&gt;
 -sleep		: How long to sleep for at startup. Default unit is seconds.&lt;br /&gt;
 -reconnect_close : Should calls be closed on reconnect?&lt;br /&gt;
 -reconnect_sleep : How long (in milliseconds) to sleep between the close and reconnect?&lt;br /&gt;
 -ringbuffer_files: How many error/message files should be kept after rotation?&lt;br /&gt;
 -ringbuffer_size : How large should error/message files be before they get rotated?&lt;br /&gt;
 -rsa		: Set the remote sending address to host:port for sending the messages.&lt;br /&gt;
 -rtp_echo	: Enable RTP echo. RTP/UDP packets received on port defined&lt;br /&gt;
 		  by -mp are echoed to their sender. RTP/UDP packets coming on&lt;br /&gt;
 		  this port + 2 are also echoed to their sender (used for sound and video echo).&lt;br /&gt;
 -rtt_freq	: freq is mandatory. Dump response times every freq calls in the log file defined&lt;br /&gt;
 		  by -trace_rtt. Default value is 200.&lt;br /&gt;
 -s		: Set the username part of the resquest URI. Default is &amp;#039;service&amp;#039;.&lt;br /&gt;
 -sd		: Dumps a default scenario (embeded in the sipp executable)&lt;br /&gt;
 -sf		: Loads an alternate xml scenario file.  To learn more about XML scenario syntax,&lt;br /&gt;
 		  use the -sd option to dump embedded scenarios. They contain all the necessary help.&lt;br /&gt;
 -shortmessage_file: Set the name of the short message log file.&lt;br /&gt;
 -shortmessage_overwrite: Overwrite the short message log file (default true).&lt;br /&gt;
 -oocsf	: Load out-of-call scenario.&lt;br /&gt;
 -oocsn	: Load out-of-call scenario.&lt;br /&gt;
 -skip_rlimit	: Do not perform rlimit tuning of file descriptor limits.&lt;br /&gt;
        	  Default: false.&lt;br /&gt;
 -slave		: 3pcc extended mode: indicates the slave number&lt;br /&gt;
 -slave_cfg	: 3pcc extended mode: indicates the file where the master and slave addresses are stored&lt;br /&gt;
 -sn		: Use a default scenario (embedded in the sipp executable).&lt;br /&gt;
 		  If this option is omitted, the Standard SipStone UAC scenario is loaded.&lt;br /&gt;
 		  Available values in this version:&lt;br /&gt;
 			  - &amp;#039;uac&amp;#039;      : Standard SipStone UAC (default).&lt;br /&gt;
 			  - &amp;#039;uas&amp;#039;      : Simple UAS responder.&lt;br /&gt;
 			  - &amp;#039;regexp&amp;#039;   : Standard SipStone UAC - with regexp and variables.&lt;br /&gt;
 			  - &amp;#039;branchc&amp;#039;  : Branching and conditional branching in scenarios - client.&lt;br /&gt;
 			  - &amp;#039;branchs&amp;#039;  : Branching and conditional branching in scenarios - server.&lt;br /&gt;
 		  Default 3pcc scenarios (see -3pcc option):&lt;br /&gt;
 			  - &amp;#039;3pcc-C-A&amp;#039; : Controller A side (must be started after all other 3pcc scenarios)&lt;br /&gt;
 			  - &amp;#039;3pcc-C-B&amp;#039; : Controller B side.&lt;br /&gt;
 			  - &amp;#039;3pcc-A&amp;#039;   : A side.&lt;br /&gt;
 			  - &amp;#039;3pcc-B&amp;#039;   : B side.&lt;br /&gt;
 -stat_delimiter : Set the delimiter for the statistics file&lt;br /&gt;
 -stf		 : Set the file name to use to dump statistics&lt;br /&gt;
 -t		 : Set the transport mode:&lt;br /&gt;
             -u1 : UDP with one socket (default), &lt;br /&gt;
             -un : UDP with one socket per call, &lt;br /&gt;
             -ui : UDP with one socket per IP address. &lt;br /&gt;
                   The IP addresses must be defined in the injection file. &lt;br /&gt;
             -t1 : TCP with one socket, &lt;br /&gt;
             -tn : TCP with one socket per call, &lt;br /&gt;
             -l1 : TLS with one socket, &lt;br /&gt;
             -ln : TLS with one socket per call, &lt;br /&gt;
             -c1 : u1 + compression (only if compression plugin loaded), &lt;br /&gt;
             -cn : un + compression (only if compression plugin loaded). &lt;br /&gt;
&lt;br /&gt;
This plugin is not provided with sipp.&lt;br /&gt;
 -timeout	: Global timeout. Default unit is seconds.  If this option is set, SIPp quits after&lt;br /&gt;
 		  nb units (-timeout 20s quits after 20 seconds).&lt;br /&gt;
 -timeout_error : SIPp fails if the global timeout is reached is set (-timeout option required).&lt;br /&gt;
 -timer_resol : Set the timer resolution. Default unit is milliseconds.  This option has an impact&lt;br /&gt;
 		  on timers precision. Small values allow more precise scheduling but impacts CPU&lt;br /&gt;
 		  usage.If the compression is on, the value is set to 50ms. The default value is 10ms.&lt;br /&gt;
 -sendbuffer_warn : Produce warnings instead of errors on SendBuffer failures.&lt;br /&gt;
 -trace_msg	: Displays sent and received SIP messages in &amp;lt;scenario file name&amp;gt;_&amp;lt;pid&amp;gt;_messages.log&lt;br /&gt;
 -trace_shortmsg : Displays sent and received SIP messages as CSV&lt;br /&gt;
 		  in &amp;lt;scenario file name&amp;gt;_&amp;lt;pid&amp;gt;_shortmessages.log&lt;br /&gt;
 -trace_screen	: Dump statistic screens in the &amp;lt;scenario_name&amp;gt;_&amp;lt;pid&amp;gt;_cenaris.log file&lt;br /&gt;
 		  when quitting SIPp. Useful to get a final status report in background mode (-bg option).&lt;br /&gt;
 -trace_err	: Trace all unexpected messages in &amp;lt;scenario file name&amp;gt;_&amp;lt;pid&amp;gt;_errors.log.&lt;br /&gt;
 -trace_calldebug : Dumps debugging information about aborted calls to&lt;br /&gt;
 		  &amp;lt;scenario_name&amp;gt;_&amp;lt;pid&amp;gt;_calldebug.log file.&lt;br /&gt;
 -trace_stat	: Dumps all statistics in &amp;lt;scenario_name&amp;gt;_&amp;lt;pid&amp;gt;.csv file.&lt;br /&gt;
 		  Use the &amp;#039;-h stat&amp;#039; option for a detailed description of the statistics file content.&lt;br /&gt;
 -trace_counts : Dumps individual message counts in a CSV file.&lt;br /&gt;
 -trace_rtt	: Allow tracing of all response times in &amp;lt;scenario file name&amp;gt;_&amp;lt;pid&amp;gt;_rtt.csv.&lt;br /&gt;
 -trace_logs	: Allow tracing of &amp;lt;log&amp;gt; actions in &amp;lt;scenario file name&amp;gt;_&amp;lt;pid&amp;gt;_logs.log.&lt;br /&gt;
 -users		: Instead of starting calls at a fixed rate, begin &amp;#039;users&amp;#039; calls at startup, and&lt;br /&gt;
 		  keep the number of calls constant.&lt;br /&gt;
 -watchdog_interval: Set gap between watchdog timer firings.  Default is 400.&lt;br /&gt;
 -watchdog_reset  : If the watchdog timer has not fired in more than this time period,&lt;br /&gt;
 		  then reset the max triggers counters. Default is 10 minutes.&lt;br /&gt;
 -watchdog_minor_threshold: If it has been longer than this period between watchdog&lt;br /&gt;
 		  executions count a minor trip.  Default is 500.&lt;br /&gt;
 -watchdog_major_threshold: If it has been longer than this period between watchdog&lt;br /&gt;
 		  executions count a major trip.  Default is 3000.&lt;br /&gt;
 -watchdog_major_maxtriggers: How many times the major watchdog timer can be tripped&lt;br /&gt;
 		  before the test is terminated.  Default is 10.&lt;br /&gt;
 -watchdog_minor_maxtriggers: How many times the minor watchdog timer can be tripped&lt;br /&gt;
 		  before the test is terminated.  Default is 120.&lt;br /&gt;
 -ap		: Set the password for authentication challenges. Default is &amp;#039;password&lt;br /&gt;
 -tls_cert	: Set the name for TLS Certificate file. Default is &amp;#039;cacert.pem&lt;br /&gt;
 -tls_key         : Set the name for TLS Private Key file. Default is &amp;#039;cakey.pem&amp;#039;&lt;br /&gt;
 -tls_crl	: Set the name for Certificate Revocation List file.&lt;br /&gt;
 		  If not specified, X509 CRL is not activated.&lt;br /&gt;
 -3pcc		: Launch the tool in 3pcc mode (&amp;quot;Third Party call control&amp;quot;).&lt;br /&gt;
 		  The passed ip address is depending on the 3PCC role.&lt;br /&gt;
 		 - When the first twin command is &amp;#039;sendCmd&amp;#039; then this is &lt;br /&gt;
 		   the address of the remote twin socket.  SIPp will try to &lt;br /&gt;
 		   connect to this address:port to send the twin command &lt;br /&gt;
 		   (This instance must be started after all other 3PCC scenario). &lt;br /&gt;
 		   Example: 3PCC-C-A scenario. &lt;br /&gt;
 		 - When the first twin command is &amp;#039;recvCmd&amp;#039; then this is &lt;br /&gt;
 		   the address of the local twin socket. SIPp will open &lt;br /&gt;
 		   this address:port to listen for twin command. &lt;br /&gt;
 		   Example: 3PCC-C-B scenario.&lt;br /&gt;
 -tdmmap 	: Generate and handle a table of TDM circuits.&lt;br /&gt;
 		  A circuit must be available for the call to be placed.&lt;br /&gt;
 		  Format: -tdmmap {0-3}{99}{5-8}{1-31}&lt;br /&gt;
 -key		: keyword value&lt;br /&gt;
 		  Set the generic parameter named &amp;quot;keyword&amp;quot; to &amp;quot;value&amp;quot;.&lt;br /&gt;
 -set		: variable value&lt;br /&gt;
 		  Set the global variable parameter named &amp;quot;variable&amp;quot; to &amp;quot;value&amp;quot;.&lt;br /&gt;
 -dynamicStart  : variable value&lt;br /&gt;
 		  Set the start offset of dynamic_id varaiable&lt;br /&gt;
 -dynamicMax    : variable value. Set the maximum of dynamic_id variable     &lt;br /&gt;
 -dynamicStep   : variable value. Set the increment of dynamic_id variable&lt;br /&gt;
&lt;br /&gt;
Signal handling:&lt;br /&gt;
&lt;br /&gt;
 SIPp can be controlled using posix signals. The following signals are handled:&lt;br /&gt;
&lt;br /&gt;
 USR1: Similar to press &amp;#039;q&amp;#039; keyboard key. It triggers a soft exit of SIPp.&lt;br /&gt;
       No more new calls are placed and all ongoing calls are finished before SIPp exits.&lt;br /&gt;
       Example: kill -SIGUSR1 732&lt;br /&gt;
&lt;br /&gt;
 USR2: Triggers a dump of all statistics screens in &amp;lt;scenario_name&amp;gt;_&amp;lt;pid&amp;gt;_screens.log file.&lt;br /&gt;
       Especially useful  in background mode to know what the current status is.&lt;br /&gt;
       Example: kill -SIGUSR2 732&lt;br /&gt;
&lt;br /&gt;
Exit code:&lt;br /&gt;
&lt;br /&gt;
Upon exit (on fatal error or when the number of asked calls (-m option) is reached,&lt;br /&gt;
sipp exits with one of the following exit code:&lt;br /&gt;
&lt;br /&gt;
           0: All calls were successful&lt;br /&gt;
	   1: At least one call failed&lt;br /&gt;
	  97: exit on internal command. Calls may have been processed&lt;br /&gt;
	  99: Normal exit without calls processed&lt;br /&gt;
	  -1: Fatal error&lt;br /&gt;
&lt;br /&gt;
Example:&lt;br /&gt;
&lt;br /&gt;
Run sipp with embedded server (uas) scenario:&lt;br /&gt;
&lt;br /&gt;
 ./sipp -sn uas&lt;br /&gt;
&lt;br /&gt;
On the same host, run sipp with embedded client (uac) scenario&lt;br /&gt;
&lt;br /&gt;
 ./sipp -sn uac 127.0.0.1&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
==See Also==&lt;br /&gt;
&lt;br /&gt;
* [[VoIP Cookbook: Building your own Telecommunication Infrastructure]]&lt;/div&gt;</summary>
		<author><name>Onnowpurbo</name></author>
	</entry>
</feed>