[Freeswitch-dev] Need some info about FreeSWITCH
Евгений Золотов
zolotov at altron.ua
Wed Aug 27 10:33:54 EDT 2008
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$" >
<action application="set" data="bypass_media=true" />
<action application="set" data="hangup_after_bridge=true" />
<action application="set" data="call_timeout=100" />
<action application="bridge" data="sofia/default/20081 at localhost:5062"
/>
<action application="sleep" data="1000" />
</condition>
</extension>
<extension name="20082">
<condition field="destination_number" expression="^20082$" >
<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 at 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 at 192.168.2.105>;tag=pavp9yj1NmH7a
To: <sip:20082 at 192.168.2.107:5062>;tag=24418SIPpTag011
Call-ID: d7bd5c78-eed8-122b-8895-00d0b7169f0d
CSeq: 103802223 BYE
Contact: <sip:mod_sofia at 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.
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