Hi Rohit,
I think you have read RFC 2833(actually this is claimed as obselete but good
to read for knowledge)/4733/4734. And guess you want to know the application
of DTMF in the context of a SIP phone because i think i saw your quey in Sip
Communicator.
An example of applcation is: say you commu
Hi All,
Could anyone please tell me what are the practical situations that a UA
(such as a SIP phone) wants to send an INVITE request with out SDP offer in
the body of the message?
Only scenario i can think of is: when clients want to know the far end full
list of media capabilities before they s
Hi,
You can receive that type of SDP. In negotiating the SDP you can accept the
media streams that are supported from your side. There's nothing like last
one replaces the previous one.
You can learn more from:
http://www.rfc-editor.org/rfc/rfc4317.txt
Hope you have already referred to RFC 3264
Hi all,
Have any of you came across implementing or studying about secure video
conferencing over SIP? if so please let me know where to look for the
sources to study about it.
Thank you in advance.
regards,
Hasini.
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Hi All,
In my sip soft phone application, I send DTMF using RFC 2976 (through sip
info message). That works fine and it is related to the SIP stack of our
application.
My questions are:
1). In implementing the DTMF sending functionality, is it a must that we
provide both implementations as in RFC
Hi,
I am developing a SIP Soft phone application for windows in C++. I need to
send DTMF as user presses keys, to the other party.
1) Is it a SIP Stack functionality or a media stack functionality?
2) Does any one knows about an API that provides application developers to
add this functionality to
> -Original Message-
> From: sip-implementors-boun...@lists.cs.columbia.edu
> [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
> Serbang, Nabam (Nabam)
> Sent: Thursday, January 15, 2009 7:34 PM
> To: BONNAERENS Ben; Hasini Gunasinghe
> Cc: sip-implementors@lists.cs.c
Hi all,
I am implementing a SIP Soft Phone application using the RTC Client API.
In both cases of initiating calls, that is; direct IP to IP and through
asterisk SIP proxy, RTC Client does not include a branch parameter in its
Via SIP header.
But the calls are connected, media exchanged and calls
