Hi Tarun,
Thanks for your quick response.
Thanks,
Kiran.
On Tue, Jun 5, 2012 at 11:39 AM, Tarun2 Gupta wrote:
> Hi Kiran
>
> Section 8.1.3.4 of RFC 3261 also states the following:
>
> "In all other respects, requests sent upon receipt of a redirect
> response SHOULD re-use the header fields an
Hi,
I don't know about Kamailio or other servers.
But to test simultaneous calls, you don't require different PC. Only you
require is different user ids and hard ware units (if you are testing media
also).
We can same port for entire sip signalling.
For media, we should use different ports for
Hi Kiran
Section 8.1.3.4 of RFC 3261 also states the following:
"In all other respects, requests sent upon receipt of a redirect
response SHOULD re-use the header fields and bodies of the original
request."
For Via branch, RFC 3261 mandates a new branch parameter, for Call-ID, To and
From, RF
Hi all,
I have a doubt regarding Cseq value while processing 3xx responses.
According to RFC 3261 8.1.3.4 the new invite request for the redirected
address can use the same Cseq value.
According to RFC 3665 3.6 call flows, the CSeq value is incremented for the
new invite request while processing t
Hi,
I want to test kamailio or any other server for it's simultaneous call
rates, form which I mean how much calls it can support simultaneously??
I know about SIPp, but that won't fulfill my purpose, because SIPp doesn't
offer simultaneity; rather it issues calls serially.
BTW, is it possible t
Ravi,
Not sure if you have raised in RFC errata, but it is already seen based on your
query.
http://www.rfc-editor.org/errata_search.php?rfc=5404&rec_status=15&presentation=table
Thanks,
Manju
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-impleme
> From: Tarun2 Gupta [tarun2.gu...@aricent.com]
>
> Hi All
>
> Consider the following scenario:
>
>
> UA 1 B2BUA UA2
> Invite (From tag F1)
> ->
> Invite (From tag F2)
>
Hi Tarun,
I have no exact idea abut SBC ,like how to check the traces as i'm new to
this SBC
I compared the 200 OK messages for successful and unsuccesful call and
there is nothing seems to be wrong.
Also in Wireshark traces we have logs for Failed and successful call (same
A party and B party)
This is a good question for Robert Sparks, who has seen most every known
behavior at SipIt.
Thanks,
Paul
On 6/4/12 7:21 AM, Kevin P. Fleming wrote:
On 06/03/2012 12:01 PM, Paul Kyzivat wrote:
OTOH, if you have no established registration state and so are doing an
initial regis
On 06/03/2012 12:01 PM, Paul Kyzivat wrote:
> OTOH, if you have no established registration state and so are doing an
> initial registration, you should be randomly be choosing a new Call-ID
> and CSeq. If*that* request fails with a 407, IMO you have some options:
> - do just as above for a retry
Hi Tarun,
According to the old saying "Be liberal with receiving and strict while
delivering", B2BUA is doing best attempt to succeed the call.
Are you sure the BYE has come for intended Dialog? (Does the Call-Id,
From-Tag, To-Tag match?).
If BYE is catching 481, it means dialog matching rules h
Hi All
Consider the following scenario:
UA 1 B2BUA UA2
Invite (From tag F1)
->
Invite (From tag F2)
-->
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