Hello,
I've been with freeswittch for a while now.. and i can say it is worth developing it.
anyhow i got into a strange issue... I'm tryng to see what load FS on my server can take. The Call flow is like this:
SIPp FS
INVITE -------->
<------- 100 Trying
<------- 302 Moved Temporary
ACK --------->
I use a dummy dialplan for that. All custom functions i've build are disabled and i'm not using it here. Also custom modules are not loaded as well.
<extension name="ServiceLookup">
<condition field="destination_number" expression="(^300030)(.*)">
<!--action application="lookup_service_destination" data="in ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, i
n $1, in
pgw01.ot.hr:5060, out red_contact, out authResult"/-->
<action application="log" data="INFO ######################## ServiceLookup ########################\n"/>
<action application="log" data="INFO ######################## contact = '${red_contact}' ##############\n"/>
<action application="log" data="INFO ######################## CallerNum = '${caller_id_number:6:16}' ##########\n"/>
<action application="log" data="INFO ######################## RADIUS auth = '${authResult}' ##########\n"/>
<action application="execute_extension" data="doRedirect XML public"/>
</condition>
</extension>
<extension name="doRedirect">
<condition field="destination_number" expression="^doRedirect$"/>
<condition field="${authResult}" expression="^0$|^60$">
<action application="log" data="INFO ######################## RADIUS auth OK!!!' ##########\n"/>
<!--action application="redirect" data="sip:${red_contact}"/-->
<!--action application="answer"/-->
<action application="redirect" data="
sip:12345616094191500@pgw01.ot.hr:5060"/>
<!--anti-action application="answer"/-->
<!--anti-action application="sleep" data="2000"/-->
<action application="hangup" data="USER_BUSY"/>
<anti-action application="redirect" data="
sip:12345616094191500@pgw01.ot.hr:5060"/>
<anti-action application="log" data="INFO ######################## RADIUS auth NOK!! ##########\n"/>
<!--anti-action application="respond" data="403 Forbidden"/-->
<anti-action application="hangup" data="USER_BUSY"/>
</condition>
</extension>
When i place a call from x-lite everything works fine ... x-lite sends an invite, gets SIP 302 and ACKs it correctly... FS is happy.
When i place a call from SIPp i have the same scenario except FS seems not understand ACK message from SIPp and re-sends SIP 302 multiple times untill it gives up.
I beleive this is due to 302 resend issue but; when i load FS with 100 CPS, i can see high CPU usage (just
one thread taking most load... the rest does almost nothing) on FS.
Also, starting from 40 CPS there is a big delay in receiving SIP 302
messages meaning i've sent 6000 calls and so far only for half of them got 302 response.
Does anybody have a clue ?
Here is a trace taken on FS for calls originated from SIPp (sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100 -trace_msg -inf test.txt -m 1 -l 4000):
freeswitch@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 16:44:26.527236:
------------------------------------------------------------------------
INVITE
sip:30003016094191500@10.4.4.251 SIP/2.0
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
Max-Forwards: 70
Contact: <
sip:22222238515000403@10.4.4.252>
To: "30003016094191500"<
sip:30003016094191500@10.4.4.251>
From: "22222238515000403"<
sip:22222238515000403@10.4.4.251>;tag=1
Call-ID:
1-6962@10.4.4.252
CSeq: 1 INVITE
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 131
v=0
o=user1 53655765 2353687637 IN IP4 10.4.4.252
s=-
c=IN IP4 10.4.4.252
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
From: "22222238515000403"<
sip:22222238515000403@10.4.4.251>;tag=1
To: "30003016094191500"<
sip:30003016094191500@10.4.4.251>
Call-ID:
1-6962@10.4.4.252
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
Content-Length: 0
------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
------------------------------------------------------------------------
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
From: "22222238515000403"<
sip:22222238515000403@10.4.4.251>;tag=1
To: "30003016094191500" <
sip:30003016094191500@10.4.4.251>;tag=Hr4mHDUeBSNyH
Call-ID:
1-6962@10.4.4.252
CSeq: 1 INVITE
Contact: <
sip:12345616094191500@pgw01.ot.hr:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
------------------------------------------------------------------------
recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
------------------------------------------------------------------------
ACK
sip:30003016094191500@10.4.4.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
To: "30003016094191500"<
sip:30003016094191500@10.4.4.251>
From: "22222238515000403"<
sip:22222238515000403@10.4.4.251>;tag=1
Call-ID:
1-6962@10.4.4.252
CSeq: 1 ACK
Contact:
sip:sipp@10.4.4.252:5060 Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070:
------------------------------------------------------------------------
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
From: "22222238515000403"<
sip:22222238515000403@10.4.4.251>;tag=1
To: "30003016094191500" <
sip:30003016094191500@10.4.4.251>;tag=Hr4mHDUeBSNyH
Call-ID:
1-6962@10.4.4.252 CSeq: 1 INVITE
Contact: <
sip:12345616094191500@pgw01.ot.hr:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063:
------------------------------------------------------------------------
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
From: "22222238515000403"<
sip:22222238515000403@10.4.4.251>;tag=1
To: "30003016094191500" <
sip:30003016094191500@10.4.4.251>;tag=Hr4mHDUeBSNyH
Call-ID:
1-6962@10.4.4.252
CSeq: 1 INVITE
Contact: <
sip:12345616094191500@pgw01.ot.hr:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
Tihomir.