We have one more problem: Step1. FreeSWITCH is started on 192.168.2.108 at sip default profile /usr/local/freeswitch/conf/sip_profiles/default.xml We remove mandatory registration of UA: <param name="auth-calls" value="false"/> Step 2. Creating at /usr/local/freeswitch/conf/dialplan/extensions $ touch 2xxx.xml and adding extension: <extension name="20081"> <condition field="destination_number" expression="^20081$" > <action application="set" data="bypass_media=true" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="call_timeout=100" /> <action application="bridge" data="sofia/default/20081@localhost:5062" /> <action application="sleep" data="1000" /> </condition> </extension> <extension name="20082"> <condition field="destination_number" expression="^20082$" > <action application="set" data="bypass_media=true" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="call_timeout=100" /> <action application="bridge" data="sofia/default/20082@192.168.2.107:5062" /> <action application="sleep" data="1000" /> </condition> </extension>

reloadxml

Step 3. We start two equal SIPp UA on 192.168.2.108 (with FreeSWITCH ) and on 192.168.2.107 in a server mode (waiting for SIP calls): $ ./sipp -sn uas -p 5062 We'll start calling SIPp UA ( in a client mode ) to both waiting for call UAs on 192.168.2.107: $ ./sipp -sn uac -s 20081 -m 1 -d 400 192.168.2.108 $ ./sipp -sn uac -s 20082 -m 1 -d 400 192.168.2.108 ==================================================================== $ ./sipp -sn uac -s 20081 -m 1 -d 400 192.168.2.108 Resolving remote host '192.168.2.108'... Done. ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(400 ms)/1.000s 5061 0.68 s 1 192.168.2.108:5060(UDP) Call limit reached (-m 1), 0.683 s period 1 ms scheduler resolution 0 calls (limit 30) Peak was 1 calls, after 0 s 0 Running, 0 Paused, 0 Woken up 0 dead call msg (discarded) 0 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg INVITE ----------> 1 0 0 100 <---------- 1 0 0 0 180 <---------- 1 0 0 0 183 <---------- 0 0 0 0 200 <---------- E-RTD1 1 0 0 0 ACK ----------> 1 0 Pause [ 400ms] 1 0 BYE ----------> 1 0 0 200 <---------- 1 0 0 0 ------------------------------ Test Terminated -------------------------------- - there is a call to UA, which located on the host where is FreeSWITCH placed and the sequence of connection and disconnection is fulfilled. $ ./sipp -sn uas -p 5062 ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Port Total-time Total-calls Transport 5062 23.86 s 1 UDP 0 new calls during 0.862 s period 1 ms scheduler resolution 0 calls Peak was 1 calls, after 7 s 0 Running, 1 Paused, 0 Woken up 0 dead call msg (discarded) 3 open sockets Messages Retrans Timeout Unexpected-Msg ----------> INVITE 1 0 0 0 <---------- 180 1 0 <---------- 200 1 0 0 ----------> ACK E-RTD1 1 0 0 0 ----------> BYE 1 0 0 0 <---------- 200 1 0 [ 4000ms] Pause 1 0 ------------------------------ Test Terminated -------------------------------- - the called side, to which FreeSWITCH makes bridge, receives all sequence of SIP messages ==================================================================== And that's 2nd experiment - the same, but the called UA (20082) placed on another host: $ ./sipp -sn uac -s 20082 -m 1 -d 400 192.168.2.108 Resolving remote host '192.168.2.108'... Done. ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(400 ms)/1.000s 5061 0.64 s 1 192.168.2.108:5060(UDP) Call limit reached (-m 1), 0.649 s period 1 ms scheduler resolution 0 calls (limit 30) Peak was 1 calls, after 0 s 0 Running, 0 Paused, 0 Woken up 0 dead call msg (discarded) 0 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg INVITE ----------> 1 0 0 100 <---------- 1 0 0 0 180 <---------- 1 0 0 0 183 <---------- 0 0 0 0 200 <---------- E-RTD1 1 0 0 0 ACK ----------> 1 0 Pause [ 400ms] 1 0 BYE ----------> 1 0 0 200 <---------- 1 0 0 0 - the causing side see SIP sequence from the FreeSWITCH side. It's OK. But the called side: $ ./sipp -sn uas -p 5062 ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Port Total-time Total-calls Transport 5062 350.51 s 1 UDP 0 new calls during 0.509 s period 1 ms scheduler resolution 0 calls Peak was 1 calls, after 12 s 0 Running, 0 Paused, 0 Woken up 0 dead call msg (discarded) 3 open sockets Messages Retrans Timeout Unexpected-Msg ----------> INVITE 1 0 0 0 <---------- 180 1 0 <---------- 200 1 9 1 ----------> ACK E-RTD1 0 0 0 0 ----------> BYE 0 0 0 0 <---------- 200 0 0 [ 4000ms] Pause 0 0 ------------------------------ Test Terminated -------------------------------- 2008-08-27 15:45:48:982 1219841148.982687: Aborting call on UDP retransmission timeout for Call-ID 'd7bd5c78-eed8-122b-8895-00d0b7169f0d'. - the called side repeats answer 200 (njrmal end of INVITE), but cann't receive ACK for it. And here UA expecting on the same UDP port, but on one host with FreeSWITCH ( i.e. another UA): $ ./sipp -sn uas -p 5062 ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Port Total-time Total-calls Transport 5062 331.42 s 1 UDP 0 new calls during 0.412 s period 1 ms scheduler resolution 0 calls Peak was 1 calls, after 6 s 0 Running, 0 Paused, 0 Woken up 20 dead call msg (discarded) 3 open sockets Messages Retrans Timeout Unexpected-Msg ----------> INVITE 0 0 0 1 <---------- 180 0 0 <---------- 200 0 0 0 ----------> ACK E-RTD1 0 0 0 0 ----------> BYE 0 0 0 0 <---------- 200 0 0 [ 4000ms] Pause 0 0 ------------------------------ Test Terminated -------------------------------- - it receives Unexpected-Msg, and that's Unexpected-Msg BYE: 2008-08-27 15:45:49:349 1219841149.349801: Dead call d7bd5c78-eed8-122b-8895-00d0b7169f0d (aborted at index 0), received 'BYE sip:127.0.0.1:5062;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.2.105;rport;branch=z9hG4bKpgDZFDN8N99gF Max-Forwards: 70 From: "sipp" <sip:sipp@192.168.2.105>;tag=pavp9yj1NmH7a To: <sip:20082@192.168.2.107:5062>;tag=24418SIPpTag011 Call-ID: d7bd5c78-eed8-122b-8895-00d0b7169f0d CSeq: 103802223 BYE Contact: <sip:mod_sofia@192.168.2.105:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.rc4-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, precondition, timer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 What do we make wrong? Also do we misunderstand in configuration FreeSWITCH.