Hello!
i have just finsished testing an asteriskserver and have some questionmarks. I ran SIPp from another computer and made calls to asterisk which played back “tt-monkey” soundfile. In SÌPp this is called calls, but in asterisk this is called channels. When we make regular calls we use sip-provider to bridge calls. This means in asterisk channels it shows as 2 channels.
So my question is:
should one interpret the SIPp result of 400 calls as 400 asterisk channels because 2 asterisk channels makes 1 call?
do you know what im trying to ask?
REgards / Marcus
A call in Asterisk is generally different than that even, as for most calls, Asterisk is not constantly playing back audio from the HDD to the calling channel/peer.
Most calls are made between two endpoints, and that’s it.
So, a call involving one channel that is connected to Background uses the most resource.
A call involving two channels with two SIP phones (UAs) talking to each other uses less resource.
Cheers.
Thanks for the reply. In your opinion, what is the best way to try performance?
I would try based upon the real-world scenario I want to emulate.
If your use case is lots of callers each calling in to hear a prompt played, then you have achieved that.
If your use case is one SIP caller talking to another, this is not the best test case for that.
Cheers.
my scenario is agents connected with Agentlogin and keeping connection to asterisk queue, feeded with calls from outside provider and also callrecording.
Okay, can you test that scenario then with sipp?
Can you create calls in to Agent login, separate calls into Queues, and the setup call recording on the Asterisk server?
Well im not that familiar with sipp, but time will tell. Right now i dont have that knowledge but google is my friend i guess 
.
/ Marcus
Help are most welcome regarding setting up SIPp for testing. i want to setup SIPp to:
*generate calls and send rtp to record on asterisk?
any ideas ?
Ok, lets say i can test asterisk by SIPp and send recorded packet with pcapplay and also record this on the asterisk. would this be equal to my real scenario where agent get call through asterisks sip-provider connection and records conversation.
/ MArcus
im having some poroblems sending pcap stream to softphone through asterisk. this sipp scenario xml is working if i run it on the asterisk server locally, but if i run it from another server on lan it doesnt work with the sipp application stream.
<?xml version="1.0" encoding="ISO-8859-1" ?>
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[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=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
Anyone out there familiar with sipp configurations that can help?
i found the problem…
had to fill -i option which i guess tells sipp what the local ip is of the server where you run the sipp test. i believe it takes 127.0.0.1 as default value if the value is not set, thats why it worked on the same machine as the astertisk runs on.
/ M