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
hi all..!!
i'm a beginner to use sipp . how can i queue calls on sipp?
thanks in advance...!!
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you can use *-m* option for the call count and *-l 1* for one call at a
time. So for example -m 20 and -l 1 will be 20 calls in the queue one call
at a time.
On Thu, Mar 31, 2011 at 3:40 PM, vibha dear vibha_de...@yahoo.co.in wrote:
hi all..!!
i'm a beginner to use sipp . how can i queue
Can anybody advice why I am getting the error in wireshark since the syntax
is correct for call hold...
On Tue, Mar 29, 2011 at 10:28 PM, Gopalakrishnan A.N sai...@gmail.comwrote:
Hi, Thanks for all your reply.
I tried with wireshark both the end. I am able find out a error through the
Hi Chandran,
I have used some scripts for Invite with call establishment. Try using this.
And change the extension number in the script file, I have used 2001. This
worked for me with my ip phone and I established a call.
On Wed, Mar 30, 2011 at 5:37 AM, mayamatakeshi
On Thu, Mar 31, 2011 at 7:56 PM, Gopalakrishnan A.N sai...@gmail.comwrote:
Can anybody advice why I am getting the error in wireshark since the syntax
is correct for call hold...
it seems you are putting two whitespaces in front of it:
v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
t=0
for making call hold in reinvite SDP put a= send only.
On Thu, Mar 31, 2011 at 4:26 PM, Gopalakrishnan A.N sai...@gmail.comwrote:
Can anybody advice why I am getting the error in wireshark since the syntax
is correct for call hold...
On Tue, Mar 29, 2011 at 10:28 PM, Gopalakrishnan A.N
a=sendonly its thereas maya said let me check for the blank spaces...
On Thu, Mar 31, 2011 at 6:11 PM, vijay kant gupta
vijaykant.it2...@gmail.com wrote:
for making call hold in reinvite SDP put a= send only.
On Thu, Mar 31, 2011 at 4:26 PM, Gopalakrishnan A.N sai...@gmail.comwrote:
use this sdp
v=0
o=alice 2890844526 2890844526 IN IP4 [media_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=sendonly
On Thu, Mar 31,
On Thu, Mar 31, 2011 at 6:28 PM, vijay kant gupta
vijaykant.it2...@gmail.com wrote:
use this sdp
v=0
o=alice 2890844526 2890844526 IN IP4 [media_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0 8 97
a=rtpmap:0
On 2011-03-31 14:41, viswavardhanreddy karna wrote:
I am using SIPp as traffic generator... in order to calculate the number
of successful calls from total number of calls which side results should
be taken?
Well, the question is, how do you define a successful call? And that's
the reason why
Hi,
Did anyone tried sending MSRP streams (TCP) using SIPp, does it work/support,
appreciate it!
Thanks
Ravi--
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Hi all,
i want to measure depending on the success-rate relevant.
i will measure server performance by sending number of calls/second to
server to load the server and by that i will be measuring the no:of
successful calls
On Thu, Mar 31, 2011 at 4:00 PM, Michael
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