Hi Sindelka, Thanks for the reply. So. All I want to do is to achieve a real flow of SIP messages to assess the performance of Asterisk using SIPp.
I have created the 8001 account in sip.conf and it's related dial plan on extensions.conf, but I haven't started the sipphone software, so It's not registered. I did a test where I removed the first line of the dial plan (exten => 8001,1,Dial(8001,20)) and tried the SIPp call, and then Asterisk replied with 404 Not Found immediately as a reply to the INVITE. My objective is to do performance testing for my Asterisk, and have it process the SIP calls for both parties (UAC, UAS). I'm running my UAS on the same machine as the Asterisk and I see that Asterisk doesn't send any INVITE or other SIP messages towards the UAS. As previously mentioned, it immediately replies to the UAC according to the configured dial plan. BTW: I have altered to the branchc.xml to have the registration messages and it did work, however, it's the same behaviour where Asterisk replies immediately without any SIP messages sent to the UAS. Can you please share a sample of the commands you run for UAC and UAS so that you force Asterisk to send SIP messages to both ? Do you run the UAC and UAS on different machies than the Asterisk machine ? Best Regards, *Yasir Saad Al-Ibrahem+1-312-428-0301* On Tue, Jul 26, 2016 at 5:10 AM, <sipp-users-requ...@lists.sourceforge.net> wrote: > Send Sipp-users mailing list submissions to > sipp-users@lists.sourceforge.net > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.sourceforge.net/lists/listinfo/sipp-users > or, via email, send a message with subject or body 'help' to > sipp-users-requ...@lists.sourceforge.net > > You can reach the person managing the list at > sipp-users-ow...@lists.sourceforge.net > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Sipp-users digest..." > > > Today's Topics: > > 1. SIPp Traffic Flow with Asterisk (yasir al-ibrahem) > 2. Re: SIPp Traffic Flow with Asterisk (sindelka) > 3. Re: Res: Res: Sipp-users Digest, Vol 53, Issue 1 (Iswarya) > > > ------------------------------ > > Message: 2 > Date: Mon, 25 Jul 2016 10:43:34 +0200 > From: sindelka <sinde...@ttc.cz> > Subject: Re: [Sipp-users] SIPp Traffic Flow with Asterisk > To: sipp-users@lists.sourceforge.net > Message-ID: <b13752d7-c202-d914-5b7a-326398732...@ttc.cz> > Content-Type: text/plain; charset="utf-8" > > Hi Yasir, > > half of your questions is actually irrelevant to SIPp :-) In the > Asterisk configuration, you've stated to continue by responding the > incoming INVITE with Ringing if the routing step > > exten => 8001,1,Dial(SIP/8001,20) > > returns with any result. So I suspect that you haven't registered any > SIP phone as extension 8001, and maybe you haven't even defined it in > sip.conf, so the Asterisk doesn't know where to send the INVITE > specified in step 1, so it returns immediately to step 2. > > Back to the SIPp setup necessary for your application: if you want to > stress-test any SIP PBX using SIPp, you need two SIPp instances: one > running the UAC scenario, i.e. acting as the calling side, where you > "modulate" the call rate and number of concurrent calls, and another one > running an UAS scenario, i.e. acting as called party, which responds all > calls with no delays (with 4xx or with 200 followed by sent or expected > BYE depending on what you want to test). So at the called side, the > embedded UAS scenario may not be enough. > > If you want to test calls to a registered local SIP user on Asterisk in > particular (because processing of such calls takes a specific path > through Asterisk's code),which seems to be your case given how you've > described it, the SIPp scenario imitating the called party must include > registration which is not the simplest task - to do that, you have to > create a pair of scenarios at the called side, an UAS one which expects > an incoming INVITE or a Cmd from the other scenario. The received INVITE > is processed normally, the received Cmd triggers sending of a REGISTER. > I've written more on this topic here at sipp-users less than a month ago. > > Also, the -ap is necessary to authorize the calling or registering > party, i.e. it must correspond to the From uri of the request sent. So > for calls *to* 8001, -ap for 8001 is useless. Plus Asterisk normally > rejects calls which come in with a locally defined number in From via a > trunk. > > As for rejection of calls coming from unknown calling parties, that's > purely an Asterisk question I don't feel competent to answer. > > Just a final remark, some time ago there was a problem that Asterisk had > first allocated all resources needed to process a call and only then it > checked whether the call was allowed, which was causing it it to hang > under floods of INVITEs. So if you manage to generate a high enough rate > of call attempts, you may have to restart the Asterisk after such test. > But it may need a farm of machines running SIPp, or not be possible at > all if the Asterisk code has changed since. > > Pavel > > > Dne 25.7.2016 v 5:49 yasir al-ibrahem napsal(a): > > Hello dears, > > > > I've been using SIPp for a week now, and I need some help to > > understand the expected message flow please. > > > > My test procedure was: > > (1) I've configured an extension on my Asterisk as '8001/8001mike' in > > sip.conf > > (2) I configured a simple dialplan for that extension so that it can > > be dialled, as: > > exten => 8001,1,Dial(SIP/8001,20) > > exten => 8001,2,Ringing() > > exten => 8001,3,Answer() > > exten => 8001,4,Hangup() > > (3) on another machine, I installed SIPp and ran it as below > > # sipp -sn uac <Asterisk IP> -s 8001 -ap 8001mike -r 1 -l 5 -m 5 > > > > Results: > > (1) calls were all successful > > (2) The call flow on Asterisk was > > --> INVITE > > <-- 100 Trying > > <-- 200 OK > > --> ACK > > --> BYE > > <-- 200 OK > > > > Questions: > > (1) Asterisk generated the above messages without issuing any actual > > messages towards the 8001 as a called party. This isn't a reliable > > stress test for Asterisk since it didn't interact with 8001. > > Is that a correct test procedure ? > > Is there a way to make SIPp traffic tests with Asterisk to generate > > SIP messages completely to both parties ? > > note: I know that If I start the sipphone (e.g. X-Lite, Zoiper) that I > > get the ringing and I must answer it manually, but I can't answer > > calls manually at a rate of 50 calls per second. > > > > (2) This is for Asterisk. > > SIPp generates INVITE messages with a caller as 'sipp@<IP>'. Do you > > know how to force Asterisk to reject INVITE messages if the caller > > isn't a registered client ? > > > > T > > ?hanks a lot? > > Best Regards, > > /Yasir Saad Al-Ibrahem > > +1-312-428-0301/// > > > > > > > ------------------------------------------------------------------------------ > > What NetFlow Analyzer can do for you? Monitors network bandwidth and > traffic > > patterns at an interface-level. Reveals which users, apps, and protocols > are > > consuming the most bandwidth. Provides multi-vendor support for NetFlow, > > J-Flow, sFlow and other flows. Make informed decisions using capacity > planning > > reports.http://sdm.link/zohodev2dev > > > > > > _______________________________________________ > > Sipp-users mailing list > > Sipp-users@lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/sipp-users > > -------------- next part -------------- > An HTML attachment was scrubbed... > >
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