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: joegen@opensipstack.org
joegen.baclor@gmail.com
Sent: Wednesday, August 13, 2008 5:27 AM
To: manoj@ascenttelecom.com; opensipstack-devel@lists.sourceforge.net
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: joegen@opensipstack.org
joegen.baclor@gmail.com
Sent: Tuesday, August 12, 2008 6:57 PM
To: manoj@ascenttelecom.com; opensipstack-devel@lists.sourceforge.net
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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