Hi,
I am developing an API, which should dump the relevant statistics that
sipp generates onto the screen.I see that the number of lines that get
generated in the stats file when I use the trace_stat option
varies.Can anyone tell me what decides the number of lines of statistics
that gets
Monica Sam -X (monsam - WIPRO at Cisco) would like to recall the message, sipp
-trace_stat option.
-
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any one know why on version sipp 1.1rc6 the following is not working ?
Run sipp with embedded server (uas) scenario:
# ./sipp -sn uas
On the same host, run sipp with embedded client (uac) scenario
# ./sipp -sn uac 127.0.0.1
I have no problem to run ./sipp -sn uas but when open another console
Amit On wrote:
Oliver,
First off all thanks for accepting both my hash function fix + the solaris
build fix.
Another thing i would like to ask from you is if you could please add My
Name/Company in the .cpp files :) maybe something like this:
* Author : Richard GAYRAUD - 04 Nov 2003
*
Monica Sam -X (monsam - WIPRO at Cisco) wrote:
Hi,
I am developing an API, which should dump the relevant statistics
that sipp generates onto the screen.I see that the number of lines
that get generated in the stats file when I use the trace_stat
option varies.Can anyone tell me what
Hi Olivier,
thanks for your reply, but the BYE is not the problem. It has a
'timeout' attribute, so SIPp is just waiting for this BYE for the time
defined in this attribute. If no BYE has been received it continues on
the label set in 'ontimeout'.
Enrico
Boulkroune, Olivier (Non-HP:Atos
I had the same problem and fixed it by adding the -i local ip address
parameter. Here is my sipp invocation that works.
sipp -sn uac -s 2005 -t un -i local ip address -r 1
Don Morrison
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Hello All,
I've attached two patches that make some minor enhancements to the latest
SIPp (11/24/06).
- retransoption.diff
This adds the ability to specify the maximum retransmission for
invites
and non-invite messages.
- countbeforeaction.diff
Count messages before
You could pause the client for 60 seconds and then go back and send the
first register message within the xml
Please ensure that you increment cseq for every register message that is
sent.
the 60 seconds value must be hard-coded (I havent found a way to accept the
value from 200 ok and use it.)
Hi,
I tried to run both SIPp 1.1rc6 and 1.1rc7 with/without IPv6 enabled on
Windows XP.
I encountered the following error:
sipp: The following events occurred:
2006-11-28 11:55:08: Unable to bind audio RTP socket (IP=::1,
port=6000), errno = 112 (Address already in use).
I read a few
I have SIPP running with an xml file derived from the built-in uas.
The only change I made is I remove the Content-Length headers where
they are hardcoded to 0. (ie, the 200OK to the BYE.)
I encountered the following problem: the end of the 200OK that SIPP
generated only had a single \r\n
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