Does IMS restricts from header to use only domain names and not IP
address.
I do not think so.
If you have seen any document mentioning it, please let us know.
Thanks
Kamal
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.c
You can send 405 Method Not Allowed, SHOULD add allow header mentioning
the methods UAS understands.
Thanks
Kamal
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Yves Dorfsman
Sent: Friday, March
maddr, sent-by, received etc.
For details I will suggest to go through RFC 3261 section 18.
Regards
Kamal
-Original Message-
From: Albert Rodriguez [mailto:rodriguez.alb...@gmail.com]
Sent: Tuesday, January 18, 2011 3:59 PM
To: Kamalakanta Palei (kpalei)
Cc: sip-implementors
uary 18, 2011 1:46 PM
To: Kamalakanta Palei (kpalei)
Cc: Brandon W. Yuille; sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] CORRECTION - SIP TCP Re Invite
withdifferent TCP Source Port
Thanks Kamalakanta,
The Re Invite in this particular session comes at 619.8 - wh
Correcting few type errors-
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Kamalakanta Palei (kpalei)
Sent: Tuesday, January 18, 2011 12:53 PM
To: Brandon W. Yuille
Cc: sip-implementors
Hi Albert
Few more inputs -
Your message transaction is as below.
(message1) > 10.200.100.1:TCP- 15000 -- Invite with SDP -->
10.200.100.2:TCP - 5060
(message2) > 10.200.100.2:TCP- 5060 -- 200 OK --> 10.200.100.1:TCP -
15000
(message3) > 10.200.100.1:TCP- 16000 -- Invite with SDP -->
10.200.100
Hi Hanifa
FQDN "p1-cscf.open-ims.test" resolve to 3 IP addresses is perfectly
fine.
Ideally you need to take the preferred IP and use.
For more details you can refer "Locating SIP servers" RFC 3263.
Thanks
Kamal
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mai
Please see response inline
-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: Friday, March 26, 2010 3:34 PM
To: Rishabh
Cc: sip-implementors@lists.cs.columbia.edu
Subject:
Standard rule applied here is -
1. You have sent INVITE with G711A.
2. In 183, you have received SDP (most probably mode=sendonly) and codec
mentioned is G711A.
Hence you are ready to receive and play G711A payload (even this is allowed
before you receive 183)
Regards,
Kamal
-Original
modifier with "X-".
And, somwhere i saw the following (forgot the source),
b=X-D: is needed for exchanging the delay
requirements.
So, it means, we can send Delay too. Is that the Delay during RTP ?
On Thu, Feb 25, 2010 at 12:36 PM, Kamalakanta Palei (kpalei) <
kpa...@cis
mention these QOS parameters during session establishment itself.
[Kamal] No.
On Thu, Feb 25, 2010 at 12:20 PM, Kamalakanta Palei (kpalei) <
kpa...@cisco.com> wrote:
>
> Not very clear about the question you are asking.
>
> Here the precondition needs to be met means , it need to
itter and delay are related to RTP. So a SDP that is part of SIP
>> INVITE cannot send them.
>>
>>
>> Regards
>> Ranjit
>>
>> -Original Message-----
>> From: sip-implementors-boun...@lists.cs.columbia.edu
>> [mailto:sip-implementor
.
On Wed, Feb 24, 2010 at 7:24 PM, Kamalakanta Palei (kpalei) <
kpa...@cisco.com> wrote:
> Hi Prem
> You can explore more on below SDP. That will give better understanding
> on precondition QoS.
>
> m=video 3400 RTP/AVP 98 99
> b=AS:75
> a=curr:qos local none
> a=cu
Hi Prem
You can explore more on below SDP. That will give better understanding
on precondition QoS.
m=video 3400 RTP/AVP 98 99
b=AS:75
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos none remote sendrecv
Thanks
Kamal
Cisco, Bangalore
India
-Original
"A" must honor BYE and send 200 OK.
Proxy "S" can send 481 or 403.
Regards,
Kamal
Cisco, Bangalore
India
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Vishal Agrawal
Sent: Thursday, December
Hi Manoj
Here I have put the example in detail how Record-Route, Route plays a
role in message routing and how it affects Req-URI.
Also a small description on Via.
Endpoint A sends a request say INVITE to proxy1, proxy1 forwards it to
proxy2 and proxy2 gives that INVITE to endpoint B.
Each proxy
What I meant by this, it looks you are trying to attempt
SUBSCRIBE - 202
SUBSCRIBE - 401
SUBSCRIBE - 200
This does not look OK.
Kamal
Cisco, Bangalore
India
-Original Message-
From: Kamalakanta Palei (kpalei)
Sent: Thursday, October 22, 2009 9:21 AM
To: 'princearim...@huawe
It is wrong to send 202 and then 401 for SUBSCRIBE.
For one request how many final responses you want to send?
Kamal
Cisco, Bangalore
India
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Prince
You can send p-myheader, ideally proxies will ignore the headers they
donot understand.
Kamal
Cisco, Bangalore
India
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
JEEVANANDHAM KARTHIC KUMAR
Sen
Please note that if INVITE passes through B2BUA, the called endpoint may
see a change in contact header.
Thanks
Kamal
Cisco, Bangalore
India
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Manoj
And probably you can add a Warning header with warning response code 399
and proper text "Event mismatch" or whatever you feel is right.
Kamal
Cisco, Bangalore
India
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbi
Why TCP connect time should be included in transaction time. The time
when you sent the message OUT transaction time starts. The 32sec is just
for SIP transaction and not inclusive of TCP connection time.
Kamal
Cisco, Bangalore
India
-Original Message-
From: sip-implementors-boun...@list
Hi Darth
>From RFC 3261 -
When comparing header fields, field names are always case-
insensitive. Unless otherwise stated in the definition of a
particular header field, field values, parameter names, and parameter
values are case-insensitive. Tokens are always case-insensitive.
So for presenc
If you make use of TLS (for send / receive of SIP messages) then you can choose
sips: otherwise you can choose sip:.
I do not think for presence one particular scheme should be used, it can be
either of two "sip:" or "sips:".
Kamal
Cisco, Bangalore
India
-Original Message-
From: [EMAI
My understanding here is you are planning to implement gateway/proxy
which can convert a SIP message to TCP message.
May be in example words "INVITE maps to TCP connect message"
If above is true, I will suggest you to map the events first then you
will be able get more insight in design and that
, 2008 11:01 PM
To: Paul Kyzivat (pkyzivat); Kamalakanta Palei (kpalei)
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Server time out - 504
Hi All,
Does anyone can tell me the reason for the error message 504 - server
time out ?
Tx
Manoj
>What is the behavior of the receiver if it receives offer with
>different
"session id" in
>re-INVITE or UPDATE?
Take down the call as REINVITE o-line with a new session id identifies a
session that does not exist at UA.
[Kamal]
Taking down the existing call may not be a good idea, one can rej
, September 26, 2008 11:41 AM
To: Kamalakanta Palei (kpalei); Bartosz Baranowski;
sip-implementors@lists.cs.columbia.edu; M. Ranganathan
Subject: RE: [Sip-implementors] In dialog error response
I think not receiving any response will trigger a timeout and automatic
dialog termination. But for a 4xx
If you receive response 408, 481 or no response then initiate dialog
termination.
Other than above cases you need not to close the dialog necessarily,
though it is implementation dependent.
If dialog was created by INVITE, dialog termination starts with sending
BYE to remote side.
Kamal
Cisc
Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iñaki Baz
Castillo
Sent: Tuesday, September 23, 2008 3:37 PM
Cc: Sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] respone code for wrong routed SIP request
2008/9/23, Kamalakanta Palei (kpalei
I hope the right response cdode is 403 Forbidden.
While proxy sends 403 to UA it indicates two information to UA.
1. Authorization wouldn't help.
2. Do not repeat the request.
Thanks
Kamal
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Klaus Darilion
S
Hi Karthik
B2BUA has got its own advantage.
Imagine a call where originating side (Say Phone A) mandates a
particular feature and the terminating side (Say Phone B) does not have
support for that. In that case B2BUA will act as terminating endpoint
for Phone A and orginating endpoint for Phone B.
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