It's also possible that the device has an outbound proxy configured
which follows strict routing. In such cases, the R-URI would be URI of
the outbound proxy which wouldn't have the user part.
-Anshuman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMA
"Now I have some people who believe that since there
is no response, the UserAgent1 should then try
BYE [EMAIL PROTECTED]>"
What will that achieve? In all likelihood, device at IP 2.2.2.2 has no
clue about the dialog between UA1 and device at IP 1.1.1.1. It makes no
sense to send BYE request t
of the first reason).
-Anshuman
-Original Message-
From: Attila Sipos [mailto:[EMAIL PROTECTED]
Sent: Friday, July 07, 2006 1:38 PM
To: Anshuman Rawat; sip-implementors@cs.columbia.edu
Subject: RE: [Sip-implementors] Multiple Contacts and BYE
I think they have a sort of proxy redundancy ide
Hi Naresh,
Tcl/Tk does not take such a long time for parsing. I have worked and
used on a SIP client which was completely written in Tcl/Tk and that
worked just as efficiently as any other soft SIP clients I have seen.
Regards,
Anshuman
-Original Message-
From: [EMAIL PROTECTED]
[mailto
That will not be a problem. All you have to do is to extract the XML
body from the SIP message and use an XML parser to parse it.
Regards,
Anshuman S. Rawat
Software Group,
Conexant Systems India Private Ltd.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf O
Sigrid,
Both are legally possible. The final result will depend on what you
actually want. Please refer to sections 13.3.1 (point 3) and 12.2.2 in
RFC 3261.
Regards,
Anshuman S. Rawat
Software Group,
Conexant Systems India Private Ltd.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[
Tadkot,
I think the problem is here --> " and insert it own VIA header field
value."
Section 9.1 of RFC 3261 states that -
"A CANCEL constructed by a client MUST have only a single Via header
field value matching the top Via value in the request being cancelled."
I am not sure about the bra
If B doesn't receive A's ACK, B will retransmit the 200 OK. From that A
will know whether B has received A's ACK or not.
Regards,
Anshuman S. Rawat
Software Group,
Conexant Systems India Private Ltd.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of rahul
S
I guess section 9.1 in 3261 answers your question -
"The following procedures are used to construct a CANCEL request. The
Request-URI, Call-ID, To, the numeric part of CSeq, and From header
fields in the CANCEL request MUST be identical to those in the
request being cancelled, including
Hi all,
I am facing a strange problem with a UA I have been working on. While
making an outgoing call the SIP proxy sends me a "SIP/2.0 404 Not Found"
response with this UA. With X-Lite UA the call goes through.
I have the INVITE message here for both calls -
My UA -
INVITE sip:[EM
from a UAC mean?
Thanks.
Regards,
Anshuman S. Rawat
Software Group,
Conexant Systems India Private Ltd.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 21, 2005 2:39 PM
To: Anshuman Rawat; sip-implementors@cs.columbia.edu
Subject: RE: [Sip-im
-
From: Jeroen van Bemmel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 21, 2005 4:09 PM
To: Anshuman Rawat; [EMAIL PROTECTED];
sip-implementors@cs.columbia.edu
Subject: Re: [Sip-implementors] SIP/2.0 404 Not Found
Anshuman,
I believe the difference in behavior comes from the difference in
See inline.
Regards,
Anshuman
2. Though an ACK is sent back to all, which one should be considered
finally
?
-> their will be only one ACK response from UAC that will the server
transaction to transition itself from completed to confirmed state, rest
of all ACK's will be absorbed by UAS having n
Here's the answer from Section 13.2.2.3 of 3261 -
"A single non-2xx final response may be received for the INVITE. 4xx,
5xx and 6xx responses may contain a Contact header field value
indicating the location where additional information about the error
can be found. Subsequent final resp
Hi,
I have a Cisco 7960 IP phone registered with Pulver (fwd.pulver.com) and
I noticed something strange with its behavior when it's registered with
Pulver.
Scenario is - There's a call between Cisco's IP phone and another IP
phone. After about 10 minutes into the call, I hang-up at Cisco phone
e
This behavior is also noticed only for long duration calls (in my case
about 10 minutes+).
A slight correction - This behavior is only noticed for long duration
calls (about 10 mins and more).
Thanks,
Anshuman
-Original Message-
From: Anshuman Rawat
Sent: Monday, January 30
Hi,
I have 2 IP phones (1 Cisco 7960 & another IP phone, say IP phone A)
registered with fwd.pulver.com. When calling Cisco from IP phone A, I
noticed that the INVITE which Cisco received that 2 equal Record-Route
values. It probably is 'cause of the fact that the INVITE contained 2
VIA field v
/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
____
From: Manjunath Warad [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 01, 2006 2:35 PM
To: Anshuman Rawat; sip-implementors
Will having a few less headers really have much affect in CPU time?
Since it is a SIP message, UAS will have to do all the things necessary
to process any SIP request.
Another point, although a 200 OK response makes sense, but practically
any response back to UAC would suggest that UAS is alive. R
ECTED]
Sent: Wednesday, February 22, 2006 4:38 PM
To: Anshuman Rawat; 'OmPrakashTripathi 70630'; [EMAIL PROTECTED]
Cc: sip-implementors@cs.columbia.edu
Subject: RE: [Sip-implementors] Determinine if a SIP device is alive.
SIP-Pinging
Check out the original SIP thread.
They are talking a
Hi All,
I have a question regarding matching server transactions. I tried to
google for the problem but couldn't find the answer.
Section 17.2.3 says
"The request matches a transaction if:
1. the branch parameter in the request is equal to the one in the
top Via header field of t
[inline]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ashish
Kumar
Sent: Thursday, May 04, 2006 3:00 PM
To: sip-implementors@cs.columbia.edu
Subject: [Sip-implementors] Query on Re-Invite
Hi,
I have question on following scenario.
A ---(Invite)
Nitin,
Your question was answered by Dale when you posted it the first time.
Quoting him -
"The above chart is incorrect -- when a proxy sends a request on, it
adds it's Via *first*. Thus, the two sets of Via's shown are reversed
from what they should be. And the two copies of the request have
>So what if, in a 302 response, you received a contact like this
>Contact:;q=0.5,;user=phone>;q=0.25
>
>So what should you do if the first contact (at 192.168.0.4) gives no
>response at all?"
I would assume that the transaction would time-out (equivalent to a 408
response) and then proceed on to
I think this might be what you are looking for -
http://www.ietf.org/rfc/rfc3960.txt
Anshuman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ???
Sent: Wednesday, June 21, 2006 11:40 AM
To: sip-implementors@cs.columbia.edu
Subject: [Sip-implementors] Mult
I believe in such a case it might be a re-transmission of an earlier
request. If it matches an existing transaction, it should re-transmit
the response.
If it doesn't, it could be a delayed re-transmitted request or a new one
(assuming that UAC generating the new request erroneously doesn't
increm
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