On Sun, Mar 20, 2016 at 8:39 PM, Tickling Contest <tickling.contest@xxxxxxxxx> wrote:
Thanks George, I think I am very, very, very confused with sipp and how it handles the coordination (I thought I knew this well, but the pause and ti. There _HAS_ to be a simpler way. It is just way. Too. Complex. I just surprised that there isn't a better tool for something that has a load of use. Maybe I should move to Asterisk based testing. Known beast...
As I said, this is what I do for more complex stuff. You have more control.
Having said that, I have used sipp to generate thousands of simultaneous TCP non-audio calls (REGISTER, OPTIONS, etc),
I have already gotten it working for a single call; you will recall in my OP I wasn't able to push it beyond about a 100 calls concurrently, and that's when I decided to let sipp manage everything.
Yeah, sorry. I forgot.
The sipp software, I think is also quite buggy.For example, I know that -p flag is supposed to take the port over which the peer registers. This port shows up when you do pjsip show endpoint <peerExtension>. I know this (i.e., -p param) works in my sipp because that's how I controlled each peer in my earlier sipp load test scenario.Well, now, the local_port does not work when I pass it as a CSV file and modify the [local_port] to [field0] etc, and as a result, the calls are not going through.They also don't have the updated documentation for release 3.5.1 which is what I am using. Sigh!
Is it possible that 3.5.1 just has too many issues? Here's what I use...
SIPp v3.4.1-TLS-PCAP-RTPSTREAM
On Sun, Mar 20, 2016 at 9:12 PM, George Joseph <george.joseph@xxxxxxxxxxxxx> wrote:_______________________________________________Oh, BTW...If sipp doesn't do it for you, there's another great tool you can use for load testing. It's called Asterisk. :)For more complex scenarios, what I've done in the past is set up 3 Asterisk instances, 1 as the call generator, 1 as the system under test, and 1 as the call receiver.On the generator instance, I have a script that keeps enough call files in /var/spool/asterisk/outgoing to simulate the number of calls I want. On the call receiver, I can set up the dialplan to do anything I want with the calls. Transfer, play something, echo, park. Whatever.On Sun, Mar 20, 2016 at 6:37 PM, George Joseph <george.joseph@xxxxxxxxxxxxx> wrote:On Sun, Mar 20, 2016 at 5:29 PM, Tickling Contest <tickling.contest@xxxxxxxxx> wrote:OK. I did that, but now, all I do is get into an infinite loop with the registrations at the callee. Here's the gist: https://gist.github.com/ticklingcontest/a0754549a88dc748f52dIdeally, here's what I need:callee registers, and accepts an infinite number of calls.caller registers, and then sends INVITES an infinite number of times so as to keep the total number of calls per the (-l parameter).It is not clear to me how I would loop at the callee scenario or caller scenario.You don't loop anything. sipp runs the scenarios itself repeatedly until -m calls have been processed.I'd start without your script or the csv files and just get a simple 1 call scenario to work.If you want some good examples, look at the Asterisk testsuite tests/channels/pjsip/basic_calls scenarios.Here's a caller file I used often...registerpause 1 secinvitepause 1 secbyepause 1 secunregisterTo simulate a call from a phone with extension/endpoint name 1100, run it like so...# sipp -sf reg_and_call.xml -s 1100 -au 1100 -ap <password> -m 1 <server:ip>If you want it to resister/call/unregister 100 times with 10 parallel calls over TCP, run# sipp -sf reg_and_call.xml -t tn -s 1100 -au 1100 -ap <password> -m 100 -l 10 <server_ip:port>Once you get that working by itself to an existing extension, set up your callee the same way, then call it from a normal working extension and make sure it responds correctly.Then have your caller call the callee, first as a single call, then try multiple calls.Only when you have that working should you introduce your injection files.What is really confusing in the caller script apart from the real confusion I have with -m and -l parameters, is how the caller's INVITE goes out from the same port as the registered port especially when they are called as two separate processes. Does sipp write a dot file somewhere where it gets its information from?Nope.BTW, In this model, I pass the CSV file that is pre-generated for the calls using a python script that looks like this:SEQUENTIALcallerID1;AsteriskIPAddress;[authentication username=silly password=sillier];calleeID1;callDuration1;callerID2;AsteriskIPAddress;[authentication username=silly password=sillier];calleeID2;callDuration2;...callerIDn;AsteriskIPAddress;[authentication username=silly password=sillier];calleeIDn;callDurationn;etc.Again, any help is appreciated. I can see how this is turning into a sipp tutorial, so I understand if you have issues dealing with this here, but I can tell that SIPp help is very sparing online.Thanks!I'll say again... If you want some good examples, look at the Asterisk testsuite tests/channels/pjsip/basic_calls scenarios. There are both inbound and outbound scenarios, authed and unauthed.
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