Re: [Sipp-users] Bug when using uac_pcap Scenario

2008-12-18 Thread martin
Hi,

Sorry for the unusual reply - I'm new to this mailing list and don't 
know how to reply to a mail in the archives.

I found the issue below from Dominique Hassel. (Maybe there are other 
posts - didn't find them yet ;-) )

I do have / found a similar problem when trying to send an audio and a 
video stream at the same time. The audio stream goes to the correct 
port, the video stream to the wrong port, because the last digit is 
truncated.

The solution suggested by Dominique solves the problem, but I have the 
weird feeling, that the problem might actually be somewhere else. I 
believe this, because this exact function seems to be called on the 
audio m-line and the video m-line. The code looks reasonable, but why 
does it work on one m-line and not on the other?

I did a wireshark-trace, but the m-lines look perfectly ok. The other 
party is XLite 3.0. I happily supply any info you need.

Any ideas / suggestions / comments?

Cheers, Martin


  Hi,
 
  I found a bug when using sipp with uac_pcap scenario. Version 3.1
 
  When sipp starts to send RTP-Traffic to my asterisk server, Wireshark
  shows an ICMP-DESTINATION-UNREACHABLE message from my asterisk.
 
  sipp -sn uac_pcap -i 192.168.1.1 -mi 192.168.1.1 -l 20 -mp 5606 -s 500
 
  extension 500 is musiconhold on asterisk
 
  I checked and found that the Destination Port Number for the
  RTP-Traffic is wrong. In the SessionDescriptionProtocol asterisk says 
  it uses Port 11994 and sipp send its RTP-Traffic to Port 1199 = The
  length of the port number is wrong.
 
  I changed in call.cpp the following line and it works.
 
  strncpy(number, begin, sizeof(number) -1 );
  to
  strncpy(number, begin, sizeof(number));
 
... rest deleted

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[Sipp-users] Can anybody give me a scenerio sample file for register?

2008-12-18 Thread Jun Yin
I want to test my firewall product with SIPP. I am going to use two PC
to work as register client and register server, the register process
should include authorization.
I searched internet but I only can find some sample for client side,
could anybody give me a pair of xml files for server side and client
side? my problem is 200OK should extract the contact info from
register packet.  but I don't know how to do it with xml.

Thanks in advance

-- 
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Web: homeofhans.homeip.net
Email: hans...@gmail.com
MSN: hans...@hotmail.com
Skype: hans_yin_vancouver

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Re: [Sipp-users] Fw: error: INT_MAX was not declared in this scope

2008-12-18 Thread Antoine
Deven,

I was the one with the orignal problem on Mandriva 2009 which I still haven't 
solved. Could you please send me a copy of your scenario.hpp file please?

I am not too sure how to include limits.h to this file, I simply added include 
/usr/include/limits.h to the file but I receive the following error:

[r...@localhost sipp.svn]# make pcapplay
make OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed 
s/Power Macintosh/ppc/` OBJ_PCAPPLAY=send_packets.o prepare_pcap.o 
PCAPPLAY_LIBS=-lpcap PCAPPLAY=-DPCAPPLAY sipp
make[1]: Entering directory `/home/guest/Desktop/SIPp/sipp.svn'
gcc   -D__LINUX -pthread  -DSVN_VERSION=\unknown\  -DPCAPPLAY -I. 
-I/usr/include/openssl  -c -o send_packets.o send_packets.c
gcc   -D__LINUX -pthread  -DSVN_VERSION=\unknown\  -DPCAPPLAY -I. 
-I/usr/include/openssl  -c -o prepare_pcap.o prepare_pcap.c
gcc   -D__LINUX -pthread  -DSVN_VERSION=\unknown\  -DPCAPPLAY -I. 
-I/usr/include/openssl  -c -o xp_parser.o xp_parser.c
g++   -D__LINUX -pthread  -DSVN_VERSION=\unknown\  -DPCAPPLAY -I. 
-I/usr/include/openssl  -c -o message.o message.cpp
In file included from sipp.hpp:63,
 from message.cpp:38:
scenario.hpp:33: error: expected constructor, destructor, or type conversion 
before string constant
make[1]: *** [message.o] Error 1
make[1]: Leaving directory `/home/guest/Desktop/SIPp/sipp.svn'
make: *** [pcapplay] Error 2

Many thanks,
Antoine





From: Deven Parekh parekh.de...@gmail.com
To: sipp-users@lists.sourceforge.net
Sent: Friday, 5 December, 2008 22:22:52
Subject: [Sipp-users] Fw: error: INT_MAX was not declared in this scope


Hi 
 
I had this error on Fedora core 9 as well.
 
It looks like the limits.h was inderectly included through 
/usr/include/bits/socket.h
 
However on Fedora Core 9,for whatever reason, the /usr/include/bits/socket.h no 
longer includes limits.h
 
Included limits.h explicitly in scenarios.hpp. Doing so seems to have got over 
the compile problem.
 
---
  DP
 
p.s: Still not sure about those  
warning: deprecated conversion from string constant to ?char*?


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Re: [Sipp-users] Problem about 2nd Via

2008-12-18 Thread Charles P Wright
You'll need two separate regexp match actions using the occurrence=1 and 
occurrence=2 tags.

Charles

Evgeny Miloslavsky emiloslav...@juniper.net wrote on 12/18/2008 01:26:52 
AM:

 Hi Giuseppe!
 As a solution/workaround for this situation I would suggest you 
 after INVITE request received by UAS, to assign first Via header to 
 variable 1 and second Via header to variable 2. While sending 
 responses from UAS put the first Via with value of variable 1 and 
 second Via will have value of variable 2.
 As far as I understand it should look like:
 recv request=INVITE
   action
 ereg regexp=.* search_in=hdr header=Via: check_it=true 
 assign_to=1,2 / 
   /action
 /recv
 
 At sending response procedure simply put the following lines:
 Via: [$1]
 Via: [$2]
 
 I hope it will help.
 
 Regards,
 
 Evgeny Miloslavsky
 Systest Engineer
 Juniper Networks Solutions Israel LTD.
 Office: 972-9-9712355 / 7320
 
 
 -Original Message-
 From: Giuseppe Roberti [mailto:j...@jnod.org] 
 Sent: Wednesday, December 17, 2008 6:04 PM
 To: sipp-users@lists.sourceforge.net
 Subject: [Sipp-users] Problem about 2nd Via
 
 Hi,
 i am testing a proxy using sipp but i have problem with Via headers.
 
 10.0.0.1 is the proxy.
 10.0.0.2 is the sipp server (-sn uas)
 10.0.0.3 is the sipp client (-sn uac)
 
 I have noticed that the 2nd via added by sipp uas is threat incorrectly
 by the proxy but i don't know if it is my fault.
 Here the sip flow.
 
  1) The sipp uac send to the proxy the INVITE:
  INVITE sip:serv...@10.0.0.1:5060 SIP/2.0
  Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK
  From: sipp sip:s...@10.0.0.3:5060;tag=23603SIPpTag091
  To: sut sip:serv...@10.0.0.1:5060
  Call-ID: 1-23...@10.0.0.3
  CSeq: 1 INVITE
  Contact: sip:s...@10.0.0.3:5060
  Max-Forwards: 70
  Subject: Performance Test
  Content-Type: application/sdp
  Content-Length:   186
  
  v=0
  o=user1 53655765 2353687637 IN IP4 10.0.0.3
  s=-
  c=IN IP4 10.0.0.3
  t=0 0
  m=audio 6000 RTP/AVP 8 101
  a=rtpmap:8 PCMA/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-11,16
  
  
  2) The proxy send the INVITE to sipp uas, adding the second Via header
  INVITE sip:serv...@10.0.0.2:5060 SIP/2.0
  Record-Route: 
sip:10.0.0.1;lr=on;ftag=23603SIPpTag091;did=c04.37fe035
  Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK21dd.7d5758a6.0;rport
  Via: SIP/2.0/UDP 
10.0.0.3:5060;rport=5060;received=10.0.0.3;branch=z9hG4bK
  From: sipp sip:s...@10.0.0.3:5060;tag=23603SIPpTag091
  To: sut sip:serv...@10.0.0.1:5060
  Call-ID: 1-23...@10.0.0.3
  CSeq: 1 INVITE
  Contact: sip:s...@10.0.0.3:5060
  Max-Forwards: 69
  Subject: Performance Test
  Content-Type: application/sdp
  Content-Length: 187
  
  v=0
  o=user1 53655765 2353687637 IN IP4 10.0.0.3
  s=-
  c=IN IP4 10.0.0.1
  t=0 0
  m=audio 50110 RTP/AVP 8 101
  a=rtpmap:8 PCMA/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-11,16
  
  
  3) The sipp uas does not properly recognize (maybe) the 2nd Via 
 and send back this (please note the line after the first Via)
  SIP/2.0 180 Ringing
  Record-Route: 
sip:10.0.0.1;lr=on;ftag=23603SIPpTag091;did=c04.37fe035
  Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK21dd.7d5758a6.0;rport, 
 SIP/2.0/UDP 10.0.0.3:5060;rport=5060;received=10.0.0.3;branch=z9hG4bK
  From: sipp sip:s...@10.0.0.3:5060;tag=23603SIPpTag091
  To: sut sip:serv...@10.0.0.1:5060;tag=15130SIPpTag011
  Call-ID: 1-23...@10.0.0.3
  CSeq: 1 INVITE
  Contact: sip:10.0.0.2:5060;transport=UDP
  Content-Length: 0
 
 Is it my fault ?
 I'm using sipp from svn.
 
 -- 
 Giuseppe Roberti
 j...@jnod.org
 
 
 
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[Sipp-users] ICMP problems

2008-12-18 Thread Elad Kugman
Hi ,

I have a lot of icmp errors during my work with sipp. It

seems like the port is close from time to time.  Do you have any
idea about this problem?



Thanks,

Elad

 

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[Sipp-users] Trouble in Paradise, sipp does not recognize incoming messages on OSX 10.5

2008-12-18 Thread TOM R
Hy people,

i started using sipp and was happy to see that it worked on Mac (thank  
you very much by the way). But the glory stopped soon because sipp can  
send packets but does not receive any. I tried sending a registration  
and waited for a 401, but nothing happened. (same with the uac/uas  
example, no receives)

According to wireshark the 401 answer arrives as it should but the  
sipp does not recognize it. I execute it with root privilege and have  
firewall deactivated so no problem there. I also tried executing in an  
Ubuntu Virtualization using Parallels, but same there.
So it does not seem to be a problem with sipp but with Leopard

I work on
Macbook Pro, SIPp 3.1 (and latest unstable) with pcapplay and ossl,  
OSX 10.5

Does anyone have an idea what  might have missed?

thanks!!!

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Re: [Sipp-users] Problem about 2nd Via

2008-12-18 Thread Giuseppe Roberti
I have used this:
  recv request=INVITE crlf=true rrs=true
   action
 ereg regexp=.* search_in=hdr header=Via: check_it=true
assign_to=1 occurrence=1/
 ereg regexp=.* search_in=hdr header=Via: check_it=true
assign_to=2 occurrence=2/
   /action
  /recv

But it doesnt work.
The variable [$2] is equal to the first variable [$1].

Thanks to all.

Charles P Wright wrote:
 You'll need two separate regexp match actions using the occurrence=1 and 
 occurrence=2 tags.
 
 Charles
 
 Evgeny Miloslavsky emiloslav...@juniper.net wrote on 12/18/2008 01:26:52 
 AM:
 
 Hi Giuseppe!
 As a solution/workaround for this situation I would suggest you 
 after INVITE request received by UAS, to assign first Via header to 
 variable 1 and second Via header to variable 2. While sending 
 responses from UAS put the first Via with value of variable 1 and 
 second Via will have value of variable 2.
 As far as I understand it should look like:
 recv request=INVITE
   action
 ereg regexp=.* search_in=hdr header=Via: check_it=true 
 assign_to=1,2 / 
   /action
 /recv

 At sending response procedure simply put the following lines:
 Via: [$1]
 Via: [$2]

 I hope it will help.

 Regards,

 Evgeny Miloslavsky
 Systest Engineer
 Juniper Networks Solutions Israel LTD.
 Office: 972-9-9712355 / 7320


 -Original Message-
 From: Giuseppe Roberti [mailto:j...@jnod.org] 
 Sent: Wednesday, December 17, 2008 6:04 PM
 To: sipp-users@lists.sourceforge.net
 Subject: [Sipp-users] Problem about 2nd Via

 Hi,
 i am testing a proxy using sipp but i have problem with Via headers.

 10.0.0.1 is the proxy.
 10.0.0.2 is the sipp server (-sn uas)
 10.0.0.3 is the sipp client (-sn uac)

 I have noticed that the 2nd via added by sipp uas is threat incorrectly
 by the proxy but i don't know if it is my fault.
 Here the sip flow.

 1) The sipp uac send to the proxy the INVITE:
 INVITE sip:serv...@10.0.0.1:5060 SIP/2.0
 Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK
 From: sipp sip:s...@10.0.0.3:5060;tag=23603SIPpTag091
 To: sut sip:serv...@10.0.0.1:5060
 Call-ID: 1-23...@10.0.0.3
 CSeq: 1 INVITE
 Contact: sip:s...@10.0.0.3:5060
 Max-Forwards: 70
 Subject: Performance Test
 Content-Type: application/sdp
 Content-Length:   186

 v=0
 o=user1 53655765 2353687637 IN IP4 10.0.0.3
 s=-
 c=IN IP4 10.0.0.3
 t=0 0
 m=audio 6000 RTP/AVP 8 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-11,16


 2) The proxy send the INVITE to sipp uas, adding the second Via header
 INVITE sip:serv...@10.0.0.2:5060 SIP/2.0
 Record-Route: 
 sip:10.0.0.1;lr=on;ftag=23603SIPpTag091;did=c04.37fe035
 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK21dd.7d5758a6.0;rport
 Via: SIP/2.0/UDP 
 10.0.0.3:5060;rport=5060;received=10.0.0.3;branch=z9hG4bK
 From: sipp sip:s...@10.0.0.3:5060;tag=23603SIPpTag091
 To: sut sip:serv...@10.0.0.1:5060
 Call-ID: 1-23...@10.0.0.3
 CSeq: 1 INVITE
 Contact: sip:s...@10.0.0.3:5060
 Max-Forwards: 69
 Subject: Performance Test
 Content-Type: application/sdp
 Content-Length: 187

 v=0
 o=user1 53655765 2353687637 IN IP4 10.0.0.3
 s=-
 c=IN IP4 10.0.0.1
 t=0 0
 m=audio 50110 RTP/AVP 8 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-11,16


 3) The sipp uas does not properly recognize (maybe) the 2nd Via 
 and send back this (please note the line after the first Via)
 SIP/2.0 180 Ringing
 Record-Route: 
 sip:10.0.0.1;lr=on;ftag=23603SIPpTag091;did=c04.37fe035
 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK21dd.7d5758a6.0;rport, 
 SIP/2.0/UDP 10.0.0.3:5060;rport=5060;received=10.0.0.3;branch=z9hG4bK
 From: sipp sip:s...@10.0.0.3:5060;tag=23603SIPpTag091
 To: sut sip:serv...@10.0.0.1:5060;tag=15130SIPpTag011
 Call-ID: 1-23...@10.0.0.3
 CSeq: 1 INVITE
 Contact: sip:10.0.0.2:5060;transport=UDP
 Content-Length: 0
 Is it my fault ?
 I'm using sipp from svn.

 -- 
 Giuseppe Roberti
 j...@jnod.org



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