Am 09.02.12 14:19, schrieb Bryant Zimmerman: > Stefan > > This is on target with my configuration I am working on. What kind of > dialplan were you using when running the tests. > Were you doing database lookups or just answering the calls and playing > hold music. Any example would be appreciated so we can quantify your test > results. I look forward to your response. > > Thanks > Bryant
the dialplan is quite simple: for the signaling up to 13500 CC i use this wait and for the 10000CC i enable the musiconhold exten => monitor,1,Noop(PERFORMANCE TESTS) exten => monitor,n,Answer ;exten => monitor,n,MusicOnHold(806,45) exten => monitor,n,Wait(45) exten => monitor,n,Hangup and i have attached the sipp scenario i was using which is also very simple. as i said i only have a 100 mbit connection on this server and its also only a virtual machine and on this physical host there are also some production machines running so i cant put it even further. very important is the rtp.conf where you have to change the rtpend port range up to 65000 or you wouldnt be able to open enough rtp ports. just to point some things out: asterisk below 1.8 was getting even slower with every version. before asterisk 1.8 was released i allready wrote some performance patches which made sip handling better and faster but the big performance boosts came too late for 1.8 but they are in asterisk 10 and the performance increase in sip udp handling is around 2 to 3 times faster. btw my normal production machines which are just the same virtual machines like this test system. i also had 330 concurrent calls, some with transcoding, many database lookups, musiconhold, pickup ... and the sysload was around 1.0 ;) best regards stefan
<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "voipgw-02.dtd"> <scenario name="Testing Subscribe"> <send> <![CDATA[ INVITE sip:monitor@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] From: voipgw-02 <sip:voipgw-02@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:monitor@[remote_ip]:[remote_port]> Call-ID: [call_id] Cseq: 1 INVITE Contact: sip:voipgw-02@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- t=0 0 c=IN IP[media_ip_type] [media_ip] m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="100" optional="true"> </recv> <recv response="180" optional="true"> </recv> <recv response="200"> </recv> <send> <![CDATA[ ACK sip:monitor@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] From: voipgw-02 <sip:voipgw-02@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:monitor@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] Cseq: 1 ACK Contact: sip:voipgw-02@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <pause milliseconds="15000"/> <send retrans="500"> <![CDATA[ BYE sip:monitor@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] From: voipgw-02 <sip:voipgw-02@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:monitor@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] Cseq: 2 BYE Contact: sip:voipgw-02@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <recv response="200"> </recv> </scenario>
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