Hi,
I had a doubt regarding the following:
If the server recieves the same INVITE request differing only in the CSeq value,
what should be the ideal behavior? Should this be treated as a new call?
THANKS
SOURAV DHAR CHAUDHURI
Get the freedom to save as many mails as you wish. To know
Hi, is valid the following escenario? All the communication is done in the TCP
connextion open by UA_1 in the first INVITE:
UA_1 UA_2
-- ---
INVITE--
---404
Hi,
Could you please tell me what should a SIP entity do if it receives a
session refresh request with Min-SE header value less than 90 seconds?
I would like to know what Error Response should be sent in this scenario.
I don't think 422 Response is appropriate for this scenario.
I'm
is that really a possible scenario.
When some UAC generates a new invite with a new CSeq why would it insert the
same From:tag and Call-ID value, which it has just sent in some previous
invite.
On Fri, Mar 28, 2008 at 12:16 PM, Sourav Dhar Chaudhuri
[EMAIL PROTECTED] wrote:
Hi,
I had a
why not use 422?
This extension introduces the 422 (Session Interval Too Small)
response code. It is generated by a UAS or proxy when a request
contains a Session-Expires header field with a duration below the
minimum timer for the server. The 422 response MUST contain a Min-SE
El Friday 28 March 2008 10:03:30 Iñaki Baz Castillo escribió:
Hi, is valid the following escenario? All the communication is done in the
TCP connextion open by UA_1 in the first INVITE:
UA_1 UA_2
-- ---
INVITE
It is perfectly valid, there are lots of such scenarios, simplest example is
call-hold.
Session refresh is one more scenario. Please go through RFC3261.
Madhav
On Fri, Mar 28, 2008 at 3:56 PM, Nitin Arora [EMAIL PROTECTED]
wrote:
is that really a possible scenario.
When some UAC generates a
I don't think 422 Response is appropriate for this scenario.
Sorry, you are probably right. 422 doesn't accurately convey the
problem.
At least with 422, they might think it's something to
with a session timer header, so hopefully they'll look
at Session-Expires and Min-SE.
This again seems to
Even Via branch can be same in case of re-INVITE?
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Hi
I m stuck in a strange problem when I send 200 OK response phone don't
respond to my call at all and when I try with some other system then it
works fine I am attaching message log of both cases please see to it n tell
me if u can help me what is that which I am missing
with regards,
Since rport (rfc3581) support worked within successful case and not the
other, rport might be required for NAT issues. And incase the UAC is
being strict, extra Via parameters such as rport should be included
within the response if they were within the request.
-Original Message-
I have seen your logs.
See, the typical scenario in offer answer exchange is
that the answer to an offer is contained in 200OK and
if caller doesn't accept that answer it sends a BYE
immediatly after receiving 200OK.
So I guess the same situation is occuring here also.
If you see the logs care
On Thu, 2008-03-27 at 16:23 -0500, Robert Sparks wrote:
From my memory of what happened at the time (which may be faulty):
After a long, and in the end relatively pointless, argument about text
vs binary formatting, the consensus of the group was to go a text
format.
Rather than
El Viernes, 28 de Marzo de 2008, M. Ranganathan escribió:
Just out of curiosity, I wonder if there are any automatically
generated parsers for the SIP ABNF out there. I can make what I did
with antlr ( which successfully made the sip parser torture tests)
available but I strongly recommend
El Jueves, 27 de Marzo de 2008, Michael Giagnocavo escribió:
software source code is written by humans, for humans (and finally for
compilers). I'm not sure anyone writes SIP messages by hand or creates them
to make them easier to read.
Even if SIP is as it is just to be human readable, it
On Fri, Mar 28, 2008 at 6:14 PM, Iñaki Baz Castillo [EMAIL PROTECTED] wrote:
El Jueves, 27 de Marzo de 2008, Michael Giagnocavo escribió:
software source code is written by humans, for humans (and finally for
compilers). I'm not sure anyone writes SIP messages by hand or creates them
to
On Fri, Mar 28, 2008 at 7:04 PM, Iñaki Baz Castillo [EMAIL PROTECTED] wrote:
Hi, AFAIK reading RFC 3261, using UDP the header Content-Length is not
mandatory and if it doesn't appear it's considered 0.
But using TCP Content-Length is mandatory. I understand that in TCP the same
connection
El Sábado, 29 de Marzo de 2008, M. Ranganathan escribió:
That would be a very very bad idea because existing implementations
are going to barf.
It's curious you say that since the fact is that many of the today existing
SIP implementations will fail if you write an horizontal TAB after the
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