[Sipp-users] sipp-3.1 is unable to play pcap files

2008-12-19 Thread Dmitry Goncharov
Hi, sipp version 3.1 has a bug in call.cpp in function get_remote_port_media(). The function improperly calculates the start position of the port. Also gcc-4.3.2 doesn't compile call.cpp and scenario.cpp since these files use INT_MAX and dont include limits.h. I am providing two patch files

Re: [Sipp-users] registration with digest auth: how to send right authorization header?

2009-01-14 Thread Dmitry Goncharov
You could try including [authentication username=your_username password=your_password] in your second REGISTER. Have a look at the enclosed file. This scenario does not register, but sets up a call through a proxy that requires digest authorization. Br, Dmitry Nikolay Kondratyev wrote: Hi

Re: [Sipp-users] registration with digest auth: how to send right authorization header?

2009-01-14 Thread Dmitry Goncharov
a standard repository… Is there an error in my scenario below? Could somebody give me a working example of registration scenario or try my scenario? Thanks and regards, Nikolay. *From:* Dmitry Goncharov [mailto:dgoncha

Re: [Sipp-users] Aborting call on unexpected message error

2009-02-12 Thread Dmitry Goncharov
Hi, Apparently, your scenario doesn't expect ACK when it comes. You should, probably, follow the sip exchange and figure out if ACK is correct. If it is correct then fix your scenario. HTH, Dmitry John Barry wrote: Hello, I want to measure the performance of a OpenSER (1.3.2)proxy doing

Re: [Sipp-users] Aborting call on unexpected message error

2009-02-13 Thread Dmitry Goncharov
I can guess that your sip server gets overloaded and doesn't answer as quickly as your scenario expects. However, this is just a guess. As is said in my previous post, you should sniff the sip exchange and figure out if ACK is correct according to SIP. If it is correct then, you should read

Re: [Sipp-users] SIPp 3.1 and 3.0 make error

2009-02-20 Thread Dmitry Goncharov
歐德旺 wrote: Dear all i had dowload SIPp-3.1 and 3.0 version ,run make or make ossl have follow error scenario.cpp:900: error: 「INT_MAX」 (在此作用欄位中尚未宣告 )was not declared in this scope my os ubuntu 8.10 dickson

Re: [Sipp-users] asterisk/sipp rtp port bug?

2009-03-14 Thread Dmitry Goncharov
E.H.Eefting wrote: After a lot of debugging with asterisk and sipp in pcap mode we found the following problem: If asterisk is configured to allow rtp ports greater than , it seems the last digit is trimmed away some how. An example of a tcpdump followed by an immediate netstat shows

Re: [Sipp-users] sendto failed with error: Address family notsupported by protocol.

2009-04-16 Thread Dmitry Goncharov
catalina oancea wrote: Hello Mike Thanks for your answer. I tried with '-t u1' and I get the same error. I also tried to run -mp 1024 and I added net.ipv4.ip_local_port_range = 1024 65000 to /etc/sysctl.conf, and then sysctl -p /etc/sysctl.conf. In my xml I use [auto_media_port], which now

Re: [Sipp-users] sendto failed with error: Address family notsupported by protocol.

2009-04-16 Thread Dmitry Goncharov
catalina oancea wrote: Hi, Thanks for the idea. What I see using strace is that normal sendto looks like this: sendto(14, \...@$\0\264z\200\0\31\345\0'\273T\21S3\17vvswwvwsvy}\376..., 180, MSG_DONTWAIT, {sa_family=AF_INET, sin_port=htons(9764), sin_addr=inet_addr(192.168.12.12)}, 16) = 180

Re: [Sipp-users] Problems creating responses in SIP-P - Im overlooking something

2009-04-24 Thread Dmitry Goncharov
Andrew Wood wrote: Hi Im trying to use SIP-P to test a SIP proxy, but the proxy keeps rejecting the 180 Ive created in SIP-P saying: Missing mandatory header fields (To, From, CSeq, Call-Id or Via) The 180 is defined as follows: send ![CDATA[ SIP/2.0 180 Ringing

Re: [Sipp-users] SIPPv2.0 question: how to send the RTP streams to a non-local IP address ??

2009-06-22 Thread Dmitry Goncharov
Huve, Frederic wrote: Folks, I'd like to know whether it is possible to use sipp to generate a SIP call to a SIP UAS, and set the media address (mi option) to an non-local IP address ? The aim is to oblige the UAS to send the RTP stream(s) only to a non local end-point (not deployed on the

Re: [Sipp-users] Only 12 simultaneous calls

2009-08-12 Thread Dmitry Goncharov
Tincho ylm wrote: Hi all! My SIPp only allow 12 simultaneous calls. If a use -l 10 everything work perfect! If I put -l 25, I get this error at 13th call: Aborting call on unexpected message for Call-Id '92-4...@ip-uac': while expecting '100' (index 1), received 'SIP/2.0 500 Server internal

Re: [Sipp-users] Issue regarding the BYE request...

2009-10-21 Thread Dmitry Goncharov
mwilliam prusty wrote: Hi All I am using SIPP unstable version.After the call is established (Exchane of INVITE,180 ringing, 200 ok,ACK between the client server ) When SIPP client is sending Bye(cseq=2), It is getting 407 from the CSCF. After that from SIPP client side i am sending the