Bin:
RFC 3863 describes application/pidf.
- Somesh
Bin Chen <[EMAIL PROTECTED]> wrote: I found the xml in the presence notify as
following:
entity="sip:[EMAIL PROTECTED]:5060">
open
sip:[EMAIL PROTECTED]:5060
online
What is the tuple id from? Or is it a don't care value? Wh
I think those timers are for un-reliable transport (UDP)
Gary Cote <[EMAIL PROTECTED]> wrote: Huseyin,
The simple answer is: yes, the client should retransmit the BYE request if
it doesn't receive a response. There are a couple of timers (Timer E and F) and
a couple of variables (T1 and T2) that
Hi,
Since P1 is B2BUA it can "remember" the transaction for some amount of time
and then check to see if it can receive any provisional responses / final
responses. On receiving such a response it can either CANCEL / BYE.
Somesh
Kasturi Narayanan <[EMAIL PROTECTED]> wrote: Hi,
Yes the Proxy can
Hi Abhishek,
Yes! UA is expected to *discard* the messages with more than one Via because
the messages might have been mis-routed.
- Somesh
abhishek goud <[EMAIL PROTECTED]> wrote: Hello
How should i discard a sip message when a sip response has more than one via
header?
Th
Hi,
I think ProxyB *should* reject the request.
Regds
Somesh S. Shanbhag
"Song, Youngsun" <[EMAIL PROTECTED]> wrote: Hi,
I have a clarification question on the following statement in Section
8.1.1.6 of RFC3261 regarding when a proxy should send a 483 and whether
the number of h
n user (TU)
--
Transaction
--
Transport
--
TU may be registrars, proxies, UA, Redirect servers, B2BUA etc.
Transaction layer for all is identified by branch_id parameter.
Hope this helps,
Regards
Somesh S. Shanbhag
Manish Jain <[EMAIL PROT
Hi Jack,
As per the RFC 3265 section 3.2.2 Subscription-State header is mandatory in
NOTIFY message. So, if Subscription-State header is not present, the endpoint
is not conforming to RFC 3515 / RFC 3265
Thanks
Somesh S. Shanbhag
"Jack W. Lix" <[EMAIL PROTECTED]> wrote: Hi,
T
Hi Young,
Have a look at the VOVIDA sip stack (http://www.vovida.org/).
It is object oriented stack and has implemented the builder and state
patterns.
Hope this helps.
Regards,
Somesh S. Shanbhag
Young Tao <[EMAIL PROTECTED]> wrote: Hi,
Where can I get a quick st
Hi,
You can have a look at OpenSER (http://www.openser.org/).
Regds,
Somesh S. Shanbhag
kirillratkin <[EMAIL PROTECTED]> wrote: Hello Gentlemen,
I'm looking for SIP Load Balancer + SIP Proxy Server solution.
Can somebody recommend such software/hardware?
(I'd
behaves as B2BUA (back to back user agent).
Thanks & Regards,
Somesh S. Shanbhag
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of
> Flex Radha
> Krishna
> Sent: Friday, July 28, 2006 9:51 AM
> To: sip-implementors@cs.columb
"
===
So, the last NOTIFY which I receive would be with no-resource and I shall
not try to re-SUBSCRIBE.
I can send the fresh REFER request for the same...
Thanks & Regards,
Somesh S. Shanbhag
Frank Shearar <[EMAIL PROTECTED]> wrote: Following on from my previo
think if the case that you have mentioned takes place I think
Transaction User (TU) must take care and Transaction Layer
shall remain un-noticed
Thanks & Regards,
Somesh S. Shanbhag
[EMAIL PROTECTED] wrote: Hi,
If the non-IST doesn't receive any RESPONSE message then
above
categories.
"
Warning 392 is miscellaneous.
Section 20.43 of RFC 3261 has some descriptive list of
warning codes which have been standardized.
Also, http://www.iana.org/assignments/sip-parameters summarises the codes.
Hope this helps,
Regards,
Somesh S. Shanbhag
Stephen Paterson <[EM
refresh the same ..?
Thanks & Regards,
Somesh S. Shanbhag
Frank Shearar <[EMAIL PROTECTED]> wrote: Say I want to transfer someone's
call. I send a REFER, establishing an
implicit subscription with the transferee. The transferee tells me
"Subscription-State: pend
Hi Ravi,
I think you can also use the SIP request - MESSAGE to intimate the balance
Its defined in RFC 3428.
Regds,
Somesh S. Shanbhag
Doug Sauder <[EMAIL PROTECTED]> wrote:
I suggest that you consider using an HTTP request from the softphone
client to a billing server. A
omesh S. Shanbhag
Markus Hofmann <[EMAIL PROTECTED]> wrote: Hi Somesh,
I have question related to this.
What is the right behaviour (sending out INVITE with SDP offer) if SDP
in 18x (non-reliable) and 2xx are different? Sending ACK / BYE? Ignoring
2xx SDP?
Regards
Markus
Somesh S Sha
treat 183 session progress as the answer and
shall ignore in subsequent responses to INVITE.
Hope this helps
Thanks
Somesh S. Shanbhag
Manpreet Singh <[EMAIL PROTECTED]> wrote: For the INVITE
transaction where the offer was sent in an INVITE and the answer coming back in
Hi,
If you are talking about SIP Timers, it is given in RFC 3261.
Chapter 17 of RFC 3261 talks in detail the use of timers in transaction layer
and the values of each timer and the meaning are given Chapter 30 of the same
RFC.
Regards,
Somesh S. Shanbhag
suvendu nayak <[EM
Hi,
I missed mentioning about SDP in previous mail.
And for SDP RFC 2327 gives the description of protocol.
and RFC 3264 gives offer answer model with SDP as applied to SIP.
Regards,
Somesh S. Shanbhag
Somesh S Shanbhag <[EMAIL PROTECTED]> wrote: Hi,
If you are talking about SIP Time
Hi,
2xx responses shall be retransmitted by the TU and not by transaction layer.
But yes, if Invite client transaction has already been terminated in "Calling"
state, then re-transmission of 2xx will continue until timeout.
But this behavior is dependent on TU.
Regards,
Somesh S
it has to wait and send
the CANCEL.
Someone, please correct me if I am wrong.
Regds,
Somesh S. Shanbhag
--- suganya <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> Suppose A calls B.
>
> B responds with 180 Ringing. But does not send 200
> ok or 200 ok gets
> lost.
>
&
Hi Ashish,
Ohhh, I thought it was UAS!
My Apologies.
Yes, the UAC *SHOULD* ignore the same.
Regds,
Somesh S. Shanbhag
--- [EMAIL PROTECTED] wrote:
>
>
> If response matches the client transaction of a
> mid-dialog request as
> per transaction matching rules(3261:17.1.3
Hi Jijo,
You can check with oSIP mailing list.
Osip mailing list
[EMAIL PROTECTED]
http://www.atosc.org/mailinglist/listinfo/osip
Regards,
Somesh S. Shanbhag
--- jijo <[EMAIL PROTECTED]> wrote:
>
> Hi All,
>
> I would like to know about the usage of SIP Baic
> Authnet
Hi Ashish,
When we are talking about SIP stacks, I shall mention
few others also.
1. oSIP stack
2. eXoSIP stack
3. VOCAL sip stack
4. Sofia sip stack (http://sofia-sip.sourceforge.net/)
5. libmsip (http://www.minisip.org/)
Regards,
Somesh S. Shanbhag
--- Banibrata Dutta <[EMAIL PROTEC
Hi Ashish,
I think it *SHOULD* return 481 Transaction/Call Leg
doesn't exist.
Regards,
Somesh S. Shanbhag
--- Ashish <[EMAIL PROTECTED]> wrote:
> Hi There,
>
> I am asking this question from SIP stack's PoV.
> On receiving a response, if transaction is
> id
Hi Muralish,
You can take the look at the following thread.
http://lists.cs.columbia.edu/pipermail/sip-implementors/2005-November/011002.html
Regards,
Somesh S. Shanbhag
--- Muralish P N <[EMAIL PROTECTED]> wrote:
> hai
> My name is Muralish.Can u please help me in
> impl
_
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
>
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>
-
SIMPLICITY IS THE BEAUTY.
BE NATURAL LIVE NATURAL.
-
Somesh S. Shanbhag
Fo
Hi Sridhar,
Some of the SIP stacks other than reSIProcate ...
oSIP
EXoSIP
JAIN Sip Stack
VOVIDA SIP Stack
Regards,
Somesh S. Shanbhag
--- Asheesh Joshi <[EMAIL PROTECTED]> wrote:
> Resiprocate is an open source free C++ SIP stack.
>
> -Original Message-
> From
ction. This is because the UAC core
handles retransmissions of the ACK, not the
transaction
layer. The ACK MUST be passed to the client transport
every time a retransmission of the 2xx final response
that triggered the ACK arrives. "
Regards,
Somesh S. Shanbhag
--- Manpreet Singh <[EM
XPIDF directly.
Regards,
Somesh S. Shanbhag
--- Kishore Sowdi <[EMAIL PROTECTED]>
wrote:
> Hi ,
> I just want to know are there any XML parsers &
> formatters available which could parse & format XML
> name spaces in the SIP message (ex:NOTIFY).
> Or shoul
transport to use while sending the messages.
Regards,
Somesh S. Shanbhag
--- Matthew Gardiner <[EMAIL PROTECTED]>
wrote:
> Thanks Somesh,
>
> Yes my stack will stamp TCP or UDP into the Via
> prior to transmitting the
> message. I was really querying which fields in the
> messa
Hi Matthew,
You have to change the Via also. Something like this
..
Via:SIP/2.0/TCP 172.16.25.17:5060;branch=z9hG4bK-c7542
Regards,
Somesh S. Shanbhag
--- Matthew Gardiner <[EMAIL PROTECTED]>
wrote:
> Hi all,
>
> Could somebody please list the part(s) of a SIP
> mes
Hi,
In this case the server has to know about the nature
of the resource wheather it is the list / single user.
Regards,
Somesh S. Shanbhag
--- Fortinsky Michael <[EMAIL PROTECTED]>
wrote:
>
> Is there a way to differentiate between
> subscriptions to a single entity
> and
to both sender and receiver side.
Regards
Somesh S. Shanbhag
--- "Ling, Fan" <[EMAIL PROTECTED]> wrote:
> Hi,
>
> The following SDP message describes a MPEG4 video
> bitstream:
>
>
>
> m=video 2 RTP/AVP 98 34
>
> a=rtpma
Hi,
Can you post the negotiated SDP's?
(SDP of request and response)
Regards,
Somesh S. Shanbhag
--- zaghouani hamdi <[EMAIL PROTECTED]> wrote:
>
> hi,
>
> i'm trying to make a call using windows messenger to
> a
> fixe phone through a gateway.
> the se
Hi Nataraju,
Yes, I think it's valid.
Regards,
Somesh S. Shanbhag
--- Nataraju A B <[EMAIL PROTECTED]> wrote:
> Hi Alll,
>
> I have a simple doubt about CSeq number in an
> out-going message
>
> Is it ok to have the same CSeq number for a REGISTER
>
Hi Kishore,
Try jain-sip-presence-proxy. It has instant messenger
(im) that subscribes with presence server for online /
offline status.
For more details take this URL
https://jain-sip-presence-proxy.dev.java.net/
Regards,
Somesh S. Shanbhag
--- Kishore Sowdi <[EMAIL PROTECTED]>
document containing the list of the
subscribers to his presence.
This also helps in acceptance / rejection of the
subscriber to watch the presence information.
Please someone correct me if I am wrong.
Regards,
Somesh S. Shanbhag
--- sudhir kumar reddy <[EMAIL PROTECTED]>
wrote:
>
Hi,
Chapter 9 of RFC 3261 describes about CANCEL.
Regards,
Somesh S. Shanbhag
--- "D. Lazreg" <[EMAIL PROTECTED]> wrote:
> Can somebody describe the CANCEL
>
> c1 -> Proxy ---c2
>
Hi,
Statemachine in reSIProcate stack might help in this
case.
Regards,
Somesh S. Shanbhag
--- Bondarenko Denis <[EMAIL PROTECTED]> wrote:
> Hi all!
> I'm looking for an example of a good SIP state
> machine (C/C++). Google
> gives a lot of links, but I'd like
Hi Udit,
Yes, I think it MUST treat it as different call.
Regards,
Somesh S. Shanbhag
--- [EMAIL PROTECTED] wrote:
> Hi,
>
> What should be the UA behaviour on receiving two
> INVITE with different
> from tags, all other headers are same,
> should it treat it as different
Hi Satyarth,
Chapter 16 of RFC 3261 describes about the Proxy
Behavior which applies to the outbound proxy also.
.. And the UA will know which will be it's outbound
proxy (through configurations).
Regards,
Somesh S. Shanbhag
--- Satyarth Negi <[EMAIL PROTECTED]> wrote:
> Hi A
MAIL PROTECTED] in a SIP Request without
> using escape characters
>
> Regards,
> // Andreas
> ___
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
>
http://lists.cs.columbia.edu/mailman/listinfo/sip-im
Hi Tauseef Hasan,
Yes, Max-Forwards is mandatory in all SIP requests.
you can check the table given in section 20 in
RFC 3261.
Regards,
Somesh S. Shanbhag
--- Tauseef Hasan <[EMAIL PROTECTED]> wrote:
> Hi
> Is it mandatory to have a Max-Forwards header field
> when a CANCEL is
Hi Aditya,
Chapter 12 of RFC 3261 speaks about strict and loose
routing.
Regds,
Somesh S. Shanbhag
--- aditya kumar <[EMAIL PROTECTED]> wrote:
> hii
> give exact definition of stricr routing & loose
> routing.how it act in a
> request give one best example.
>
though we get NOTIFY with body
PUA1 MUST NOT rely on the body
This can be configured in the Presence Server in the
form of accept / deny lists.
Please correct me if I am wrong.
Regards,
Somesh S. Shanbhag
--- sudhir kumar reddy <[EMAIL PROTECTED]>
wrote:
>
> Hi All,
>
&g
n the case of an INVITE).
Generally, the host portion of this URI is the IP
address or FQDN of the host. The URI provided in the
Contact header field MUST be a SIP or SIPS URI."
Someone correct me if I am wrong.
Regards,
Somesh S. Shanbhag
--- Jaya krishna <[EMAIL PROTECTED]> wrote:
server
IP and port details.
Regards,
Somesh S. Shanbhag
--- "Bharat M. Sarvan" <[EMAIL PROTECTED]>
wrote:
> Hello everybody,
>
>I have coded a program which
> processes the Subscribe
> request and sends back a 200 ok response. I have
>
Hi,
I don't think there is some limitation in size in
bytes
of To-tag and From-tag.
Please someone correct me, if I am wrong.
Regards,
Somesh S. Shanbhag
--- Guy Schreiber <[EMAIL PROTECTED]> wrote:
> Hi Experts,
>
> Is there any limitation to the size in bytes of th
RTP/RTCP traffic.
You can try jrtplib as well.
RFC 3550 talks about RTP and its header format.
Regards,
Somesh S. Shanbhag
--- rajesh kumar <[EMAIL PROTECTED]> wrote:
> I am connecting to the server through clinet socket
> program written in C.
> I am sending INVITE request and I
@cs.columbia.edu
>
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>
---
SIMPLICITY IS THE BEAUTY.
BE NATURAL LIVE NATURAL.
---
Somesh S. Shanbhag
Mascon Global Communication Technologies
Enterprise of Mascon Global Limited
#59/
Hi Martin,
I think it's valid.
You can check section 17.2.1 of RFC 3261.
Regds,
Somesh S. Shanbhag
--- Martin Koenig <[EMAIL PROTECTED]> wrote:
> Hello all,
>
> is it legal per RFC3261 to send two separate "100
> trying" provisional
> responses for a SIP R
Hi Rajesh,
Could you please explain me "Message build error"?
Also Check with INVITE wheather you are sending SDP
with it. If you aren't sending the SDP through INVITE,
you MUST in ACK, because you have to convey the
codec's and connection information to destination.
Regds
an automatically reply with some 403 Forbidden
message.
This is possible from looking at original identity of
the sender of the message (From header in the
request).
Regds,
Somesh S. Shanbhag
--- journal sateesh <[EMAIL PROTECTED]>
wrote:
> Hi all
> I want to know how this access
Hi Bharat,
RFC 3265 describes SUBSCRIBE and NOTIFY messages.
RFC 3665 describes about basic call flows.
Regds,
Somesh S. Shanbhag
--- Somesh S Shanbhag <[EMAIL PROTECTED]> wrote:
> Hi Bharath,
>
> You can refer to RFC 3665 which describes SUBSCRIBE
> and
> NOTIFY messag
t
the From URI not contain IP addresses or the FQDN of
the host on which the UA is running, since these are
not logical names.
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
ip-implementors
>
-----------
SIMPLICITY IS THE BEAUTY.
BE NATURAL LIVE NATURAL.
---
Somesh S. Shanbhag
Mascon Global Communication Technologies
Enterprise of Mascon Global Limited
#59/2, 100Ft
ng 401
Authentication required (User to User Authentication
is allowed).
We can include Authorization header in the request
(INVITE) (for 401).
Regards,
Somesh S. Shanbhag
--- Shikhar Sarkar <[EMAIL PROTECTED]> wrote:
> Guys,
>
> Is there a way to authenticate a SIP use
Hi,
UAS may reject the request with 421(Extension
Required)
If it is so, it MUST contain Require: header.
Regds,
Somesh S. Shanbhag
--- Attila Sipos <[EMAIL PROTECTED]> wrote:
>
> Technically, it's not supported or required
> (No "timer" in Require or Supp
here is a
delay in generating the response, the UAS SHOULD add a
delay value into the Timestamp value in the response.
This value MUST contain the difference between the
time of sending of the response and receipt of the
request, measured in seconds.
>>>>>>>>>>>>>>
60 matches
Mail list logo
