RFC or 3GPP standard but not
yet able to find any.
Your expert inputs are much appreciated and if it can be provided in
some reference to standard it will be very helpful.
--
Thanks,
Varun Bhatia
___
Sip-implementors mailing list
Sip-implemen
anything.
Thanks,
Varun
On Wed, Oct 15, 2014 at 7:52 PM, Sourav Dhar Chaudhuri
wrote:
> Hi,
> Can CRBT works without using Reliable Provisional Response ?
>
>
>
> A INVITE (with SDP offer) > B
>
> A <===
Regards,
Varun Bhatia
___
Sip-implementors mailing list
Sip-implementors@lists.cs.columbia.edu
https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
ip to check the correct
one for sending the responses.
Sent from Iphone,
Varun
> On 19-Sep-2014, at 22:35, Kchitiz Saxena wrote:
>
> No Vivek, its definitely not NAT. It is 2 IP addresses in DNS server for a
> hostname.
>
> On Fri, Sep 19, 2014 at 7:50 PM, Vivek Talwar
Thanks Ankur, not sure if it is implemented in such manner then we should not
honor new connection with INVITE?
Thanks,
Varun Bhatia
> On 29-Apr-2014, at 21:05, ankur bansal wrote:
>
> Hi Varun ,
> I dont think sip protocol will have any issue with this .Means you can use
> on
Thanks Brett, is there any specific standard which indicates that INVITE
dialog will be using same connection of REGISTER ?
Thanks,
Varun
On Tue, Apr 29, 2014 at 4:35 PM, Brett Tate wrote:
> RFC 5626 will likely be helpful.
>
> > -Original Message-
> > From: sip
handling between REGISTER
request and INVITE request ?
Any inputs are appreciated.
--
Regards,
Varun Bhatia
___
Sip-implementors mailing list
Sip-implementors@lists.cs.columbia.edu
https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
t;with a BYE request, using the dialog identifier retrieved from the
>
>2xx final response.
>
>
>
>
>
> Best Regards,
>
> Vivek Batra
>
>
>
>
>
>
>
> *From:* VARUN BHATIA [mailto:varuninbha...@gmail.com]
> *Sent:* Friday, April 11, 20
equest as described in Section 15."
>
> Best Regards,
> Vivek Batra
>
>
> -Original Message-
> From: sip-implementors-boun...@lists.cs.columbia.edu
> [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of VARUN
> BHATIA
> Sent: Friday, April
each represents a distinct
dialog, with a distinct dialog identifier.
The UAC core MUST generate an ACK request for each 2xx received from
the transaction layer.
Any inputs are really appreciable.
Thanks,
Varun Bhatia
___
Sip-implementors mailing
0 RTP/AVP 34
>
> port = 0, changing the RTP codec to a known static one (H263).
>
>
> Regards.
>
> ___
> Sip-implementors mailing list
> Sip-implementors@lists.cs.columbia.edu
> https://lists.cs.columbia.edu/mailman/listinfo/sip-im
Hi,
I have received:
a=path:msrp://10.166.46.226:7664/CR003+1382409119.486157738;tcp
My stack is rejecting it, not sure about the precise grammar for it, seems
that this may be due to + sign received in it.
Any inputs are appreciable.
Thanks,
Varun
--
Regards,
Varun Bhatia
Hi,
I have received:
a=path:msrp://10.166.46.226:7664/CR003+1382409119.486157738;tcp
My stack is rejecting it, not sure about the precise grammar for it, seems
that this may be due to + sign received in it.
Any inputs are appreciable.
Thanks,
Varun
Hi,
I would really appreciate if somebody could answer the below questions.
Regards
varun
--- On Wed, 12/15/10, varun wrote:
> From: varun
> Subject: [Sip-implementors] Publish method for presence server
> To: sip-implementors@lists.cs.columbia.edu
> Date: Wednesday, December 15,
rify this.
Regards
Varun
___
Sip-implementors mailing list
Sip-implementors@lists.cs.columbia.edu
https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
happen in this case.
Regards
Varun
___
Sip-implementors mailing list
Sip-implementors@lists.cs.columbia.edu
https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
Hi,
When a SIP UA sends an Invite with Authorization credentials( username and
password) to Proxy, how does the proxy authenticate the username/password, I
mean how does the proxy know if it wants to allow that particular user to make
a call.
Appreciate your response.
Thanks
varun
It would be really helpful if somebody could reply to
this.
Thanks
Varun
--- varun <[EMAIL PROTECTED]> wrote:
> Hi,
> The expires header field specifies the time duration
> for which the request is valid( lets say INVITE).
> Can i specify the Expires header field greater th
3xx,4xx response, cant ACK go end to end
if the UAS has specified its Contact header field...
Please clarify.
Thanks
Varun
--- Bob Penfield <[EMAIL PROTECTED]> wrote:
>
> The ACK for 2xx is a separate transaction because it
> needs to go end-to-end.
> ACK for 3xx-6xx are h
terminated in 32 secs).
What if expires is less than 32 secs( 64 *T1)and no
final response is received during that time, who will
terminate the transaction (UAS or UAC)...will UAS send
some 4xx response in this case?
Thanks
varun
My first query still remains unanswered, would be glad
if somebody could put across his thoughts.
Thanks
Varun
1>In case of Invite-200 OK, why is ACK a separate
transaction while for any other non2xx final response,
ACK is a part of the same transaction.
--- [EMAIL PROTECTED] wr
ctions?Does it forward it upstream to UAC?
What does a 6xx response exactly mean( global
failure??..
3> When a UAC transaction times out, it generates a
408 locally. Is there a case in which 408 is send from
the UAS/PROXY??
I would appreciate if somebody could answer my queries
ASAP.
Tha
Could somebody reply please~!
--- varun <[EMAIL PROTECTED]> wrote:
> Hi,
> I tried to understand the use of privacy and
> P-asserted Identity headers in SIP but not really
> clear about these.
> Could somebody explain in simple terms the use of
> these two headers.
Hi,
I tried to understand the use of privacy and
P-asserted Identity headers in SIP but not really
clear about these.
Could somebody explain in simple terms the use of
these two headers.
Need this info ASAP, would be really helpful if
somebody could reply.
Regards
varun
oes it mean no media path in both directions or same
as case 1( one way media from A->B)/.
Please reply ASAP.
Thanks
varun
--- Andrea Rizzi <[EMAIL PROTECTED]> wrote:
> There are no violations at all in responding with
> inactive, and as you
> mentioned this will suspend any me
?
Thanks
varun
Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news,
photos & more.
http://mobile.yahoo.com/go?refer=1GNXIC
___
Hi,
Where exactly is tel URL used, how is a tel URL
converted into a SIP URL.Please give some details.
Thanks
varun
Be a better Globetrotter. Get better travel answers from someone who knows.
Yahoo
Hi,
What actually is a dynamic Payload type in SIP.Can
somebody please explain?
Thanks
Varun
Luggage? GPS? Comic books?
Check out fitting gifts for grads at Yahoo! Search
http://search.yahoo.com/search
Hi,
If Cancel is send to a proxy for an Invite which has
been forked by the proxy, will the cancel also get
forked and send to the differenet UAS.
Thanks
varun
--- Patel Nishant-JTPD86 <[EMAIL PROTECTED]> wrote:
> Hi Varun,
>
> An INVITE that is forked can cause multiple
Varun
Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news,
photos & more.
http://mobile.yahoo.com/go?refer=1GNXIC
___
Could anybody reply please??
Thanks in advance
varun
--- varun <[EMAIL PROTECTED]> wrote:
> Hi,
> Another media issue:
>
> user A->GateWay->user B
>
> --->Invite
> < 200 OK
>
> Ack is lost( no Ack)
>
> Here the G
Hi,
I am talking about a case in which SIP side starts
sending media but the other end of call is not ready
yet to play media.So what should the gateway do?
Buffer the media or just drop the media packets!!
Thanks
Varun
--- Vishal Mathur <[EMAIL PROTECTED]> wrote:
> Hi Varun,
>
>
e get an ACK.
Thanks
varun
Shape Yahoo! in your own image. Join our Network Research Panel today!
http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7
___
media.
Please suggest.
Regards
varun
Luggage? GPS? Comic books?
Check out fitting gifts for grads at Yahoo! Search
http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+g
How will B2BUA ensure that it remain in call
siganlling path without adding the record route??Could
you explain please?
Thanks
Varun
--- A B <[EMAIL PROTECTED]> wrote:
> A B2BUA adding a record-route does not make sense.
>
> Consider the below scenario, a call from A to B via
35 matches
Mail list logo