Re: [OpenSIPStack] GIPs integration to OpenSIPStack

[Manoj K. J. <ma...@as...>]

Actually in my opinion i will have to do more than codec inclusion. I am not
sure if you have seen GIPS functions..they way they use is...

(When Softphone is Registered)
	- We create an instance of GIPS using Initialize function.
(When a new call is initiated i.e. INVITE is about to sent)
	- We create a new channel
	- We specify RTP listen port (Same is sent in SIP SDP)
(When Session progress comes)
	- We start listening to RTP port.
	- We start playing incoming media.
	- We set the "Send IP" and "Send port" (That comes in session progress SDP)
(When 200 OK comes)
	- We start sending RTP to Mediaproxy IP and port
(Hangup)
	- We close all channels.

I have included a PDF file with this email. In page 31 there is a table
which explains above points more. In Page 29 there is provision to add
Encryption scheme.

As i understand i might need to touch more than codec part in opensipstack
but you will know better than me.



-----Original Message-----
From: jo...@op... [mailto:joe...@gm...]
Sent: Wednesday, August 13, 2008 5:27 AM
To: ma...@as...; ope...@li...
Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack


Manoj Kumar Joshi wrote:
> Thanks for Replying joegen. I am on it already. What about ...
> 1 - Starting Audio devices?
> 2-  Start/Stop RTP
>

Both these two should be seamless after you have successfully registered
your codec.  Unless, what you want is to rewrite everything and start
from scratch?
> I also need to understand how encryption is implemented as i would need to
> encrypt RTP also using same functions right? If i get an overall picture
it
> would be real easy for me to device some flowchart on paper and proceed
with
> work.
>

Encryption is implemented in rtp.cxx and Encryption.cxx

RTP_Session::SendReceiveStatus RTP_UDP::ReadDataOrControlPDU()
BOOL RTP_UDP::WriteData()
BOOL RTP_UDP::WriteControl()

This too should be seamless after you introduced your codec.
> Regards,
>
> Manoj
>
> -----Original Message-----
> From: jo...@op... [mailto:joe...@gm...]
> Sent: Tuesday, August 12, 2008 6:57 PM
> To: ma...@as...; ope...@li...
> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack
>
>
> The first thing you need to do is to implement your codec as a subclass
> of OpalFramedTranscoder.  You can check out how Speex is implemented
> (speexcodec.h, speexcodec.cxx) and base you custom codec from there.
> The next step is to call your codec registration macro in allcodecs.h.
>
> Manoj Kumar Joshi wrote:
>
>> Hello,
>>
>> I am looking forward to integrate GIPS media processing to opensipstack.
>> Initially i want to incorporate it only on its Softphone interface and
>>
> build
>
>> ATLSip with it.
>>
>> I think i would need to make changes in SDP, Audio devices handling,
>> Start/stop RTP and encryption. I tried to find some documentation on
>>
> openSBC
>
>> architecture but did not find much. Please give me some of your valuable
>> directions on how should i start with this.
>>
>> Regards,
>>
>> Manoj
>>
>>
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