Re: [Sipp-users] Dead call (successful) error

-- Michael Hirschbichler, Mag. Dipl.-Ing. Institute of Telecommunications Vienna University of Technology A-1040 Wien, Favoritenstr. 9-11/388 Phone: +43 1 58801 38846 -- This SF.net email is sponsored

[Sipp-users] SIPp-wish :)

Hi all, I am currently developing another SIPp-XML-scenario and I have one *wish* for future functionalities: Today, if I want to analyse the response codes of e.g. a failed call, it looks this way: recv response=400 optional=true

Re: [Sipp-users] Simple to configure SIP Proxy

-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Michael Hirschbichler, Mag. Dipl.-Ing. Institute of Telecommunications Vienna University of Technology A-1040 Wien, Favoritenstr. 9-11/388 Phone: +43 1 58801 38846

Re: [Sipp-users] SIPp 3.4 roadmap

Hi Rob, mentioning the error-log: can you take a look at the time-formatting of the -trace_msg logs? Currently, they are formatted as 2013-02-16 23:40:21:589.216 which is incorrect. The correct (parseable) formatting should (IMHO) be 2013-02-16 23:40:21.589216 (as the ms und us are fractions

Re: [Sipp-users] SIPp 3.3

Hi Rob, fine job - from my point of view, 3.3b2 works fine. I use it on a Raspi (compiled under Raspbian for ARM) and Ubuntu for continuous voice quality testing and I did not notice any problem. Plans and wishers for 3.4 release? Well, it would be phantastic, if sipp can create a log file

Re: [Sipp-users] SIPp 3.3

-authentication responses - here we monitor surprisingly high processing latencies. The funny thing is: the subsequent 407 challenge (in a new SIPp instance, to an INVITE request) is handled deterministic without any spikes. br Michael Josh -Original Message- From: Michael Hirschbichler

Re: [Sipp-users] SIPp 3.3 beta 2

Hi, First of all: thanks for reanimating the SIPp-project and the planned reactivating of the sipp-wiki :) I successfully built sipp 3.3b2 on a raspberry with rasbian for my continuous VoIP-quality testing project. I built it with make pcap_ossl and make all - except a load of the common

Re: [Sipp-users] Loadtesting Asterisk with Sipp through Kamailio

Hi, this won't work AFAIK - sipp sends requests only to one remote socket during a call. br Michael Am 07.09.2012 08:06, schrieb Grant Bagdasarian: Hello, I’m trying to perform a loadtest on our Asterisk machines using Sipp, but there is a SIP Proxy(Kamailio) in between. Kamailio acts as a

Re: [Sipp-users] sipp play pcap file question ,can you help me?

do to play the two file same time. tks for your answer ,thank you! -- Michael Hirschbichler, Mag. Dipl.-Ing. Institute of Telecommunications Vienna University of Technology A-1040 Wien, Favoritenstr. 9-11/388 Phone: +43 1 58801 38846

Re: [Sipp-users] unable to parse the msg field

I don't think, that this tag is working. try this: recv request=INVITE optional=true br Michael Am 2011-04-14 09:36, schrieb manasi: recv request=INVITE optional=true -- Benefiting from Server Virtualization:

Re: [Sipp-users] SIPp Wiki

I have the same problem as well. In beginning of 2011, the wiki was reachable, but outdated and spammed. I contacted Olivier, but also got no response. Got the impression, hp is not more interested in SIPp :( br Michael Am 2011-04-11 09:51, schrieb wondra: Greetings, I am using SIPp in a

Re: [Sipp-users] successfull calls for server performance

at acks with regards, viswavardhan On Thu, Mar 31, 2011 at 3:13 PM, Michael Hirschbichler s...@hirschbichler.biz mailto:s...@hirschbichler.biz mailto:s...@hirschbichler.biz mailto:s...@hirschbichler.biz wrote: On 2011-03-31 14:41

Re: [Sipp-users] successfull calls for server performance

I propose you take the KPIs from RFC 6076 - Basic Telephony SIP End-to-End Performance Metrics, br Michael On 2011-03-30 23:23, viswavardhanreddy karna wrote: Hi every one, I have a doubt regarding the calculation of server perforrmance. Should we take successfull

Re: [Sipp-users] successfull calls for server performance

, 2011 at 9:55 AM, Michael Hirschbichler s...@hirschbichler.biz mailto:s...@hirschbichler.biz wrote: I propose you take the KPIs from RFC 6076 - Basic Telephony SIP End-to-End Performance Metrics, br Michael On 2011-03-30 23:23, viswavardhanreddy karna wrote: Hi

Re: [Sipp-users] DTMF pcap files with payload type=127

Hi, untested: create a SIPp-(UAC)Scenario with a SDP containing only the a common Voice-pt (like 0 or 8) and the pt=127 as you want. Then start Wireshark, run the scenario and call a SIP Hardphone directly. On this hardphone push the buttons in the wished order. In your wireshark-Trace, you

Re: [Sipp-users] Use of SIPP along with kamailio - REGISTER followed by INVITE not working

Two questions: * the 36000 expiration - is it acknowledged by the Registrar (See Contact-Header in the 200OK response)? * Are you behind a NAT? BR Michael On 2011-01-04 11:24, Stephen McVarnock wrote: Hi, I have got the second scenario here to work i.e. REGISTER xml ran, kill sipp, run sipp

Re: [Sipp-users] Subscribe/Notify Call Flow

:4085551...@10.253.205.110:5060 Allow: INVITE,ACK,BYE,CANCEL,PRACK,NOTIFY,INFO,OPTIONS,UPDATE,SUBSCRIBE Supported: 100rel, timer,precondition Message-Waiting: yes Anything else you need please do tell. Cheers, Rawat On Thu, Apr 22, 2010 at 21:40, Michael Hirschbichler s

Re: [Sipp-users] Subscribe/Notify Call Flow

and worked though with some errors. Will get back if not resolved + more Questions. Cheers, Rawat On Fri, Apr 16, 2010 at 17:46, Michael Hirschbichler s...@hirschbichler.biz mailto:s...@hirschbichler.biz wrote: Well, this is not tricky, just send SUBSCRIBE... /send recv response=200

Re: [Sipp-users] Subscribe/Notify Call Flow

Well, this is not tricky, just send SUBSCRIBE... /send recv response=200/recv recv request=NOTIFY/recv send 200 OK ... /send For the SIP-messages, just start the SIP-client of your choice and record the messages with wireshark br Michael On 2010-04-16 13:55, Himanshu Rawat wrote: Hi

Re: [Sipp-users] crazy problem on simple call scenario

This scenario as described below won't work. If I understood the description correctly, the signalling-flow is UA Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- --INVITE- In sipp, the mapping of a message (request/reply) is done by parsing for the SIP Call-ID -

Re: [Sipp-users] about sipp capacity

Hi, 408cps (SIP only, no media) are definetly no problem for a standard UAS-scenario on a current hardware-configuration. Furthermore, a UAS does not generate error-responses automatically by itself (except a BYE-response if the -nd - flag is not set), br Michael WANG Jin jia Jw wrote: To

Re: [Sipp-users] SIPp with Open IMS Core, OK message does not arrive from IMS core

Hi, Your Call-ID is 1-17...@127.0.0.1, where a call-id in sipp in general is composed of 'number of call'-'processid'@'local_ip'. So, IMHO, sipp takes the wrong, the loopback, IP-address for the variable [local_ip]. Try sipp again and use the -i-parameter to explicitely define the IP adress

Re: [Sipp-users] successful dead call issue

Cool, SIPp, IMS and IPv6 ;) Hmm, my guess is, that the ACK is not recognized correctly by the IMS-core-logic and the dead-call messages are retransmissions. As first debugging step, I propose to add a pause of a few seconds at the end of the UAC-scenario. Then, you will see, if there are any

Re: [Sipp-users] SIPp can't send RTP pckts to AST

IIRC there was a substr-Bug in earlier sipp-versions. Try updating to the newest version, br Michael srt_liyq schrieb: 1. SIPp sends Invite（with sdp） 2. Receive the Response Message 200 OK from AST， including Media Description, name and address : audio 10178 RTP/AVP 8 101. 3.

Re: [Sipp-users] 407 for INVITE

OK, imho: obviously, the call setup succeeded after beeing challenged by your x-cscf - this means, the challenge in the 407 - Response was interpreted correctly and used for the 2nd INVITE-request by the sipp uac. The x-cscf accepted the credentials and forwards the INVITE-request to the

Re: [Sipp-users] 407 for INVITE

Are you sure, the 407 is for the second INVITE with the credentials? Are you getting a 100/180/183/200 for the second INVITE-request? My guess is, that your ACK for the 407 is not correct and the subsequent 407ers are retransmissions. br Michael PS: please no x-posting. mwilliam prusty wrote:

Re: [Sipp-users] UAS not quitting after reaching call-limit

Hi all, am I the only one having this problem? Does anyone know how to solve this issue? br Michael (Xposted and follow-up set to devel-list) Michael Hirschbichler wrote: Hi all, I am currently using sipp30 (SIPp v3.1-TLS-PCAP, version svn522M) and when running as UAS, it does

[Sipp-users] UAS not quitting after reaching call-limit

Hi all, I am currently using sipp30 (SIPp v3.1-TLS-PCAP, version svn522M) and when running as UAS, it does not automatically quit after receiving -m-Requests. Another sipp-Version (SIPp v2.0.1-TLS-PCAP, version 20070516) is quitting correctly. So, well, the question: is it a bug, or a feature?

[Sipp-users] Understanding RTP in SIPP

Hi all, I am using a pcap sound sample to be replayed with sipp with nop action exec play_pcap_audio=./mediastream.pcap/ /action /nop This media-stream in the pcap-file has some jitter and various inter-packet-delay. Am I correct in the assumption, that this jitter and

Re: [Sipp-users] Understanding RTP in SIPP

it arrives? That's going to be days, weeks, or months in this case. Cheers, Todd. -Original Message- From: Michael Hirschbichler [mailto:s...@hirschbichler.biz] Sent: Tue 10/20/2009 3:56 AM To: sipp-users@lists.sourceforge.net Subject: [Sipp-users] Understanding RTP in SIPP Hi

[Sipp-users] Starting sipp paused?

Hi all, I want to start multiple sipp instances in parallel. To synchronise them, I am planning to use the remote control UDP-socket. So, I want to start each instance one after another in the paused state and then I want to send each of them the p-letter to start the traffic as synchronised as

Re: [Sipp-users] Starting sipp paused?

this behaviour built in, with its manager process controlling the SIPp instances. See http://sipp.sourceforge.net/ims_bench Regards, -David -Original Message- From: Michael Hirschbichler [mailto:s...@hirschbichler.biz] Sent: mercredi 16 septembre 2009 14:21 To: sipp-users

Re: [Sipp-users] sendto failed with error: Address family not supported by protocol.

catalina oancea wrote: I also tried with snapshot http://sipp.sourceforge.net/snapshots/sipp.2009-01-21.tar.gz. The same problem occurs. The sipp command is: /usr/local/sipp//sipp -sf scen.xml -t un -r 20 -l 200 -aa -i 192.168.13.13 -m 1000 -inf cases.csv -trace_rtt -trace_screen

[Sipp-users] Bug in -bind_local parsing?

Hi all, I noticed a strance behaviour when passing an -bind_local - Argument: Following the online-help: ./sipp -h the remotehost must be added as first argument: sipp remote_host[:remote_port] [options] entering ./sipp 2.2.2.2 -sn uac -bind_local 1.2.3.4 results in ---

Re: [Sipp-users] [ sipp-Patches-1823593 ] raw sockets for spoofing source IP address/port

Hi all! I wanted to use this patch, but am I correct, that it is currently not merged with the main tree? I also tried to patch the diff against trunk-rev. 332 (as used in the diff-file : --- sipp.hpp(revision 332) +++ sipp.hpp(working copy) Surprisingly, also against rev. 332, the

Re: [Sipp-users] Asterisk and Authoriation

Just increase the CSEQ-Number of the 2nd INVITE (message 5), BR Michael d 82 k schrieb: Hi everybody, I would like to test my asterisk and in order to do this I would like to run sipp on two computers (A and B) and register some users (1001 : 1010 for A and 2001 : 2010 for B) and make

Re: [Sipp-users] SIPp discards if it receives extra header in the response

You have to let SIPp generate the Call-ID: --- [call_id] A call_id identifies a call and is generated by SIPp for each new call. In client mode, it is mandatory to use the value generated by SIPp in the Call-ID header. Otherwise, SIPp will not recognise the

[Sipp-users] using Retry After:-Header

, who posted their SIPp-cps - high-scores! I think, Charles Wright made the race with 10.000 calls per second. -- Michael Hirschbichler, Dipl.-Ing. Institut fuer Breitbandkommunikation Technische Universitaet Wien A-1040 Wien, Favoritenstr. 9-11/388 Tel: +43 1 58801 38846

[Sipp-users] SIPp Performance - High-Score?

the performance - or the type - of the SUT, but just the current cps - high-score, You ever archived with SIPp. I think, this fact would also be useful for the wiki, thanks in advance and BR Michael -- Michael Hirschbichler, Dipl.-Ing. Institut fuer Breitbandkommunikation Technische Universitaet Wien A-1040

[Sipp-users] fieldn in authentication-header

Hi all! I just updated SIPp to the most current version, but as I wanted to run my scenarios, I noticed, that the auth-error is back again: the line [authentication username=[field4] password=[field1]] creates as a

Re: [Sipp-users] Question: Which VoIP-testools do You use?

Michael Hirschbichler wrote: Hi all, I am working on an overview about different VoIP/SIP/IMS - related testtools, and I am wondering, which tools(GPL and non-free) are You using. Well, some tools I already found during my recherche: * The great and wonderful SIPp ;-)) * Protos-Test

Re: [Sipp-users] Question: Which VoIP-testools do You use?

Michael Hirschbichler wrote: Hi all, I am working on an overview about different VoIP/SIP/IMS - related testtools, and I am wondering, which tools(GPL and non-free) are You using. Well, some tools I already found during my recherche: * The great and wonderful SIPp ;-)) * Protos-Test

[Sipp-users] [branch]-struggle

: [last_header_field] or [last_header_index_field], like [last_Via_0_brach]? Best regards Michael -- Michael Hirschbichler, Dipl.-Ing. Institut fuer Breitbandkommunikation Technische Universitaet Wien A-1040 Wien, Favoritenstr. 9-11/388 Tel: +43 1 58801 38846

[Sipp-users] Listen to more than one UDP-Socket?

Hi! I want to use sipp in the current version to listen to more than one local UDP-Port. Is this possible in some way? Background: I register 1500 different user from the same host, but with different port-numbers in the Contact:- and the Via:-Header field. After registering, I create 1500

[Sipp-users] [fieldn] int the [authentication ...]-part

Hi all! I am trying to inject data from a .csv-file into the [authentication ...]-line of a REGISTER-request. Am I correct in the assumption, that this isn't working in the current rc8? Example: The xml-line [authentication username=[field0] password=[field1]] creates: Authorization: Digest

Re: [Sipp-users] Question

Hi I didn't test your XML-file, but you have at least one bug: The correct syntax is recv response ... /recv send ... #SIP code ... /send Your /recv-tag is located at the end of the xml-snipplet and that's wrong :-) Greets Michael Federico La Volpe wrote: Hi guys, I am new on this