arate-transaction-for-invite-with-a-successful-200-ok%C2%A0response%C2%A0/
5. Will see other to respond on logical reason behind it.
Thanks
Vivek Batra
On Wed, Dec 30, 2015 at 12:02 PM, Karthik.v wrote:
> Hi all,
>
> I have some list of queries on sip ,
>
>
>
> 1
AFAIK, both of the flows are incorrect. In first case, if SDP offer is in
reliable provisional response, PRACK must contain SDP answer. UPDATE can be
used any time once SDP offer answer has been done in provisional response
and PRACK.
Best Regards,
Vivek Batra
On Fri, Dec 18, 2015 at 6:15 PM
aybe they are standardized by 3gpp.)
>
> Thanks,
> Paul
>
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> __
.
Similar cases also exist during call park etc.
Best Regards,
Vivek Batra
On Wed, Dec 17, 2014 at 10:06 AM, Seshagiri Kondaveti <
seshagiri.kondav...@radisys.com> wrote:
>
>
> Hi all,
>
> What is the use case for Inivite without SDP ?
> I see most of the softphones sends out i
Alok/Srinivas,
Agree
Best Regards,
Vivek Batra
On Thu, Dec 4, 2014 at 2:55 PM, Alok Tiwari wrote:
> Hi Vivek,
>
> Here the issue is media is not in sync. If UAS is not providing the SDP in
> answer, how UAC can ensure whether the ongoing media is reliable anymore
> and if the
Hi all,
Just thinking widely if it's violating the SDP offer/answer in ReINVITE,
then why to terminate the complete dialog instead of clearing only specific
transaction...
Best Regards,
Vivek Batra
On Thu, Dec 4, 2014 at 2:42 PM, Alok Tiwari wrote:
> Hi Tarun,
>
> IMO, th
y get
transferred.
Does it answer your question or do you mean that in your case, user agent B
is not able to decode (or doesn't support) Refer-To?
Best Regards,
Vivek Batra
On Tue, Oct 28, 2014 at 6:07 PM, Sourav Dhar Chaudhuri <
sourav_mi...@yahoo.co.in> wrote:
> Hi Vivek,
>Th
d by INVITE message.
Best Regards,
Vivek Batra
On Tue, Oct 28, 2014 at 4:35 PM, Sourav Dhar Chaudhuri <
sourav_mi...@yahoo.co.in> wrote:
> Hi,
> Is this mandatory to have support for replaces parameter to support
> REFER request?
>
> Means If Supported: replaces
Johan,
I was also thinking on sending reinvite, but it will only change the media
IP address and won't create the new dialog to transfer target...
Best Regards,
Vivek Batra
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors
held call?
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 Sander
Rambags
Sent: Friday, May 09, 2014 1:07 PM
To: sip-implementors@lists.cs.columbia.edu
Subject: [Sip-implemen
receiving
subsequent 2XX, it must be able to match 2XX using dialog identifier and
generate ACK accordingly followed by BYE.
Best Regards,
Vivek Batra
From: VARUN BHATIA [mailto:varuninbha...@gmail.com]
Sent: Friday, April 11, 2014 12:04 PM
To: Vivek Batra
Cc: sip-implementors
Subject: Re
ing early dialog associated
with the request. If the SIP entity receives a subsequent 2xx final
response, it will normally generate and send an ACK request, followed
with a BYE request, using the dialog identifier retrieved from the
2xx final response.
Best Regards,
Vivek Ba
hat dialog, then the UAC MUST terminate the dialog
by sending a BYE request 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
BHAT
itial mail, since initial INVITE is not having
any To tag value, in result CANCEL should also not include To tag value
whereas CANCEL message shown below includes To tag. IMO, if CANCEL and
INVITE should be same, To tag should not be included in CANCEL.
Best Regards,
Vivek Batra
-Original Message
w and suggest what is going
wrong here.
Best Regards,
Vivek Batra
INVITE sip:011441157180@192.168.1.136:5060 SIP/2.0
From: "Anonymous"
;user=phone;tag=ccid-176140755-358439439
Contact: sip:NO_CALLER_ID@69.132.136.5:5060
Via: SIP/2.0/UDP 69.132.136.5:5060;branch=z9hG4bK-358439439-1
message, I can
see To tag coming in CANCEL message from service provider.
Do you think UAS should ignore the To tag and complete the CANCEL
transaction?
Best Regards,
Vivek Batra
Request-Line: INVITE sip:011441157180@192.168.1.136:5060 SIP/2.0
From: sip:1714627@69.132.136.5:5060;user
within dialog say BYE,
Request-URI will be set to 'sip:1...@abc.com' or 'sip:proxy2.com'? This is a
case where proxy2 follows strict routing.
Will appriecate your comments.
Best Regards,
Vivek Batra
___
Sip-implementor
default 3600 seconds), that binding
(IP1) shall remain live in the registrar server. Any proxy module requesting
registrar sever for the location information of UA gets two binding viz IP1
and IP2 however IP1 nowhere exists in the network. Any solution to take care
of this?
Best Regards,
Vivek
Hi,
Prior to RFC 3264, presence of port=0 in the SDP was also used as an
indication to put the call on hold.
> I have never seen such real time implementation (may be I have ignored
if it exists). To place the call on hold, either use the media attribute
'SendOnly' (RFC 3264) or set the conn
llow header. If RFC
2833 negotiation fails and remote end hasn't capability of detecting
DTMF in SIP INFO, you haven't any other choice than sending DTMF in
Inband :)
Different people may have different views but that is what implemented
in most of the live products.
Best Rega
Hi Iñaki,
Thank you very much for sharing this information. I check the same in my
server implementation with various SIP clients and hope this should work
out.
Best Regards,
Vivek Batra
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors
server
can be send in the certificate and subjectAltName should resolve this
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 Iñaki
Baz Castillo
Sent: Monday, June 13, 2011
Certificate during SIP
TLS call when server is connected behind the NAT router
On 06/13/2011 08:54 AM, Iñaki Baz Castillo wrote:
> 2011/6/13 Vivek Batra:
>> However, when IP-PBX is connected behind the NAT router with private IP
>> address assigned on its Ethernet interface (SIP cl
have ever come to know about
such problem..
Best Regards,
Vivek Batra
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have placed my query in SIP forum.
Best Regards,
Vivek Batra
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Hi,
I have found it with several service providers which play announcement (and
hence send 183 Session Progress) to wait if actual called party is busy and
send 180 Ringing as soon as called party becomes available.
Best Regards,
Vivek Batra
-Original Message-
From: sip-implementors
ved domain from SRV for which DNS A query was done,
in URI part of SIP message?
Best Regards,
Vivek Batra
-Original Message-
From: Iñaki Baz Castillo [mailto:i...@aliax.net]
Sent: Wednesday, May 19, 2010 7:56 PM
To: Victor Pascual Avila
Cc: Vivek Batra; sip-implementors@lists.cs.columbia.
ned in NAPTR response) OR domain for which NAPTR query was
done in URI part of SIP message?
Best Regards,
Vivek Batra
-Original Message-
From: Iñaki Baz Castillo [mailto:i...@aliax.net]
Sent: Wednesday, May 19, 2010 7:56 PM
To: Victor Pascual Avila
Cc: Vivek Batra; sip-implement
Thanks Dale for the reference. I will refer RFC 3263 for my query and come
back in case of any issue.
Best Regards,
Vivek Batra
-Original Message-
From: WORLEY, Dale R (Dale) [mailto:dwor...@avaya.com]
Sent: Wednesday, May 19, 2010 12:57 AM
To: Vivek Batra; sip-implementors
UAC and domain resolved using DNS SRV query
is xyz.123.com, which domain should be used in SIP message viz xyz.com or
xyz.123.com?
I would highly appreciate your suggestions.
Best Regards,
Vivek Batra
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Sip-impl
layed locally only if provisional response is not having SDP answer.
Best Regards,
Vivek Batra
-Original Message-
From: Sunita Bhagwat [mailto:sunita_bhag...@infosys.com]
Sent: Tuesday, March 02, 2010 5:53 PM
To: Vivek Batra; sip-implementors@lists.cs.columbia.edu
Subject: RE: [Sip-impleme
should be played only when called device is ringing.
I believe that gateway should get 180 Ringing (after 181 Call is being
forwarded) when called mobile phone starts ringing and RINGING message is
received from GSM network. Once this occurs, then only RBT should be played.
Best Regards,
Vivek
an OG call or should be used during both Outgoing and Incoming Call?
Thanks and Kind Regards,
Vivek Batra
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ired
since rport will help only in getting responses across NAT, whereas to
receive further transaction request, UA needs to know and publish its public
IP address/port.
Best Regards,
Vivek Batra
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Iñaki,
Just wanted to confirm about the use case you defined! In either case where
Bob accept or deny the new SDP offer in Re-INVITE, where the use case exists
for CANCEL?
Best Regards,
Vivek Batra
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip
Have you been confirmed whether SIP UDP Ports are not blocked by ISP? This
is very common.
Try to use some other UDP port which probably should not be blocked by ISP.
Best Regards,
Vivek Batra
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip
Ofcourse not an empty rtp stream. If your IPPBX is having B2BUA, I will
suggest to play Music on Hold (MoH) in RTP packets to the SIP provider.
Best Regards,
Vivek Batra
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun
is, caller can
dial the desired number to reach specific extension.
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 Vivek
Batra
Sent: Thursday, September 03, 2009
n only 180 Ringing should be honored to play RBT locally else
RTP from ITSP should be played".
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
Avasarala Ranjit-A20990
th contact in Register and you
can expect the same header in contact of 200 OK, this will make the work
easy to fetch the expiry timer for individual binding from 200 OK.
Best Regards,
Vivek Batra
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-impleme
To implement Busy Lamp Field (BLF), you should refer RFC 4235.
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 rishabh
Sent: Wednesday, August 26, 2009 12:53 PM
Hi
The below is valid scenario.
Also RFC 3261 section 13.2.1 mentions
"The UAC MUST treat the first session description it receives as the answer,
and MUST ignore any session descriptions in subsequent responses to the
initial INVITE."
[Vivek] - But that is not the case since 180 Ringing has n
>> Greetings,
>> I am wondering if the below scenario is valid or not.
>>
>> <-- 183 (with SDP) then,
>> <-- 180 (without SDP)
>
> Yes it's.
> However it depends in UAC behaviour on how to render it to the human
> (it could choose to render the early-media comming from the same 183,
> or it could c
port only from IP Address recieved in SDP answer from called party
(not assumed any NAT scenerio :))
Actually what happens in absence of this restriction, my SIP client start
mixing the RTP packets recieved from any IP address!
Any suggestion??
Best Regards,
Vivek Batra
: Dale Worley [mailto:dwor...@nortel.com]
Sent: Saturday, May 09, 2009 3:00 AM
To: Vivek Batra
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] Early dialog can be replaced if
TransferTarget is the reciepient of dialog (early) during Attendant Call
Transfer
On Wed
are
following this), we shall also have to ignore such statements from RFC.
I would appreciate you valuable feedbacks if you have been came across such
issues.
Best Regards,
Vivek Batra
-Original Message-
From: Vikram Chhibber [mailto:vikram.chhib...@gmail.com]
Sent: Thursday, May 07
states to not
replace the early dialog if not initiated by the Transfer Target. Transfer
Target in this case is the recipient of the dialog (early).
Best Regards,
Vivek Batra
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I agree with Ranjit/ ibc. This can be observed in various IPPBX which
responds the SUBSCRIBE with 403 Forbidden when event is recognized but not
supported for specific users (subscribers) due to limitation of resources
etc.
Best Regards,
Vivek Batra
Message: 5
Date: Thu, 2 Apr 2009 12:36
state of Alice itself which is actual
called party instead maintaining the *dialog* (from NOTIFY perspective)
state of its own UAS core.
Best Regards,
Vivek Batra
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Sip-implementors@lists.cs.columbia.e
SIP client on other call leg does not
support INFO?
I was afraid by blocking the INFO since it can also be used to transport
other information.
Best Regards,
Vivek Batra
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Sip-implementors
implementation by VoIPtalk? (However, call gets placed
successfully by Fwd.Pulver).
Thanks and Kind Regards,
Vivek Batra
-Original Message-
From: [EMAIL PROTECTED] on behalf of Vivek
Batra
Sent: Tue 8/19/2008 5:42 PM
To: sip-implementors@lists.cs.columbia.edu
Subject: [Sip-implementors] Digest
Hi,
I have query regarding challenge response mechanism (digest authentication,
MD5) in SIP as follows:
A and B are SIP clients registered with B2BUA.
A calls B and sends INVITE to B2BUA. B2BUA challenges INVITE with response
407 Auth.
A again sends the INVITE with authentication header (say
identify its own binding to get the
expiry timer allocated by registrar to UA2.
--Vivek
_
From: Rockson Li (zhengyli) [mailto:[EMAIL PROTECTED]
Sent: Friday, June 27, 2008 3:42 PM
To: Vivek Batra; Scott Lawrence; sip-implementors@lists.cs.columbia.edu
Subject: RE: [Sip-implem
same hostname while registering from
multiple location (same as in case of IP address even private).
By considering the same, adding randomly generated tag should be better
option!!
Best Regards,
Vivek Batra
-Original Message-
From: Rockson Li (zhengyli) [mailto:[EMAIL PROT
have same IP:Port.
Best Regards,
Vivek Batra
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