gards,
> Somesh
>
> -Original Message-
> From: sip-implementors-boun...@lists.cs.columbia.edu [mailto:
> sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext goutam
> kumar
> Sent: Wednesday, September 22, 2010 3:15 PM
> To: sip-implementors@lists.cs.columbia.ed
Hi,
I'm trying to implement a VOIP call between two endpoints. I'm in a doubt.
Say Alice and Bob are in a call. Now,
STEP I
Alice puts Bob on hold. i.e.
INVITE (RTP-sendonly)
Alice -> Bob
200 OK (RTP-recvonly)
<--
Hi vinod,
There is an open source SIP stack available called osip. It is in C and you
can find more about it at
http://www.antisip.com/as/
there is also a layer implemented over it by the name eXosip. It provides a
sort of an application layer over osip. Do check on the licensing issues if
any, i
t only to enforce the reliability
of the 200 OK??
Thanks & Regards,
Goutam
On Tue, Jul 6, 2010 at 11:51 AM, Alex Balashov wrote:
> On 07/06/2010 02:04 AM, goutam kumar wrote:
>
> > 1) If for an incoming call, the callee dosen't receive an ACK for a 200
> OK,
> > s
Hi,
During implementation I have come across two situations,
1) If for an incoming call, the callee dosen't receive an ACK for a 200 OK,
should the call fail??
2) In an outgoing call, if the end-user hangs-up a call, i.e. goes ON_HOOK
while the remote party has still not answered the call, shoul
Hi,
During implementation I have come across two situations,
1) If for an incoming call, the callee dosen't receive an ACK for a 200 OK,
should the call fail??
2) In an outgoing call, if the end-user hangs-up a call, i.e. goes ON_HOOK
while the remote party has still not answered the call, shoul
Hi,
thanks a lot...
On Tue, Jun 29, 2010 at 6:59 AM, WORLEY, Dale R (Dale) wrote:
>
> From: sip-implementors-boun...@lists.cs.columbia.edu [
> sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of goutam kumar
> [gottybl...@gmail.com]
Hi,
In case we receive an INVITE without an SDP body, as per rfc 3261, it is
mentioned that the 200 OK from our side should contain the offer-SDP and the
ACK from the calling party will contain the answer-SDP.
But, in such case, how do we alert the user agent at the receiving end and
then generat
Hi Tarun,
The record-route header field is actually a list of addresses. It is
mandatory that every request or response between two UAs MUST pass via all
the addresses in that list.
So, if an incoming INVITE has a record route header then any response to it
MUST have the same header in it, so that
Yeah you are right. I was trying to cite an example where the Require header
is used(which in my example, was related to the initial INVITE or
reINVITE method and not the MESSAGE method)
On Thu, Jun 24, 2010 at 5:33 PM, Brett Tate wrote:
> > The require header field is generally added to let
> >
Hi Ranjit,
The require header field is generally added to let the other party know
about your requirement.
Say for example, for call establishment you need an SDP answer in the
provisional 1xx response, i.e you require PRACK support.
In that case, you will send 100rel in the 'require' header field
un 24, 2010 at 4:19 PM, Joegen E. Baclor
wrote:
> On 6/24/10 6:09 PM, Frank Shearar wrote:
> > On 2010/06/24 11:07, goutam kumar wrote:
> >
> >> Hi,
> >>
> >> I am trying to implement the offer-answer model as per RFC 3264. When I
> went
> >
odecs
simultaneously throughout the call in order to be able to receive data of
any of the two types. Am I right??
Thanks,
Gotham
On Thu, Jun 24, 2010 at 3:39 PM, Frank Shearar <
frank.shea...@angband.za.org> wrote:
> On 2010/06/24 11:07, goutam kumar wrote:
> > Hi,
> >
&g
Hi,
I am trying to implement the offer-answer model as per RFC 3264. When I went
through RFC 4317 to see the various scenarios of this negotiation I found
this:
" 2.2. Audio and Video 2
Alice can support PCMU, PCMA, and iLBC codecs, but not more than one
at the same time. Alice offers al
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