Open Source Sip: [OpenSIPStack] GIPs integration to ...

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2 Replies Last post: Aug 13, 2008 9:35 PM by Guest

Guest

Aug 13, 2008 6:29 AM

[OpenSIPStack] GIPs integration to OpenSIPStack

Guest

1. Aug 13, 2008 6:29 AM

in response to: Guest

Re: [OpenSIPStack] GIPs integration to OpenSIPStack

I guess there is one way to use non-gips codecs also. I will do some more
research and answer you. Please also answer following .

From what places Encryption related functions are being called (Like enable

encrytion, then encrypting and decrpting all SIP messages) Also if i
understand correctly threre is some key involved in order to encrypt RTP.

Best Regards,

Manoj

-----Original Message-----
From: joegen@opensipstack.org joegen.baclor@gmail.com
Sent: Wednesday, August 13, 2008 3:42 PM
To: manoj@ascenttelecom.com; opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack GIPs integration to OpenSIPStack

If GIPs has its own RTP and media input/output channels, would GIPS be
willing to share it with other codecs as well if SDP negotiation ended
up with a none-GIPS codec? If not, then you wouldn't be able to re-use
the SoftPhone classes and instead implement your own subclass of
CallSession and CallSessionManager. If GIPS RTP and Audio Channels are
exposed to other codecs, then it would be doable by merely replacing
RTP_UDPSession with the GIPS counterpart and a GIPS wrapper to
PSoundChannel.

Manoj Kumar Joshi wrote:

Yes

-----Original Message-----
From: joegen@opensipstack.org joegen.baclor@gmail.com
Sent: Wednesday, August 13, 2008 2:26 PM
To: manoj@ascenttelecom.com
Cc: opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack GIPs integration to OpenSIPStack

GIPS has its own RTP Stack, audio input/output channels?

Manoj Kumar Joshi wrote:

Dear Joegen,

GIPS is not only a set of few codec rather it has its own media

processing

system. The functions i specified in last email need to be called from
various points of opensipstack. If you can direct me from where all media
processing is done i will be able to understand better.

In addition to that please also tell me following...

From what places Encryption related functions are being called (Like

enable

encrytion, then encrypting and decrpting all SIP messages) Also if i
understand correctly threre is some key involved in order to encrypt RTP.

you can plaese give me complete flow it will be a great help.

Best Regards,

Manoj

-----Original Message-----
From: joegen@opensipstack.org joegen.baclor@gmail.com
Sent: Wednesday, August 13, 2008 1:37 PM
To: manoj@ascenttelecom.com
Cc: opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack GIPs integration to OpenSIPStack

Manoj Kumar Joshi wrote:

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.

As far as i can tell this will be done for you by the OpenSIPStack
subsystem. Just register your codec to it. I do not know anything
about GIPs but if its the same as all other codecs which basically has
the ability to expose, encode and decode methods, then all you need is
plug it in.

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.

I was under the impression that you are referring to the XOR encryption
of opensipstack. If its another proprietary encryption other than srtp,
then you will have to give more info how it works and probably I could
point you to the right place where to hack it in OpenSIPStack.

-----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

encrypt RTP also using same functions right? If i get an overall

picture

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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Guest

2. Aug 13, 2008 9:39 PM

in response to: Guest

Re: [OpenSIPStack] GIPs integration to OpenSIPStack

Firing up file search in visual studio for "Engine::Encrypt"

C:\var\home\oss\devel\opensipstack\src\SIPMessage.cxx(8963): return
Encryption::Engine::Encrypt( in, out );
C:\var\home\oss\devel\opensipstack\src\SIPUDPTransport.cxx(346):
Encryption::Engine::Encrypt( in, out );
C:\var\home\oss\devel\opensipstack\src\opal\src\rtp\rtp.cxx(2201):
Encryption::Engine::Encrypt( frame, out );
C:\var\home\oss\devel\opensipstack\src\opal\src\rtp\rtp.cxx(2244):
Encryption::Engine::Encrypt( frame, out );

By process of elimination, I would say SIPUDPTransport is the place you
should be looking at for SIP Encryption. The key for encryption is a
static member variable of Encryption::Engine::m_Key. Just set it to the
value you like and off you go.

Manoj Kumar Joshi wrote:

I guess there is one way to use non-gips codecs also. I will do some more
research and answer you. Please also answer following .

From what places Encryption related functions are being called (Like enable

Manoj Kumar Joshi wrote:

GIPS has its own RTP Stack, audio input/output channels?

Manoj Kumar Joshi wrote:

Dear Joegen,

GIPS is not only a set of few codec rather it has its own media

processing

From what places Encryption related functions are being called (Like

enable

encrytion, then encrypting and decrpting all SIP messages) Also if i
understand correctly threre is some key involved in order to encrypt RTP.

Manoj Kumar Joshi wrote:

Actually in my opinion i will have to do more than codec inclusion. I am

not

SDP)

(When 200 OK comes)
- We start sending RTP to Mediaproxy IP and port
(Hangup)
- We close all channels.

opensipstack

but you will know better than me.

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

encrypt RTP also using same functions right? If i get an overall

picture

would be real easy for me to device some flowchart on paper and proceed

with

work.

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

This SF.Net email is sponsored by the Moblin Your Move Developer's

challenge

Build the coolest Linux based applications with Moblin SDK & win great

prizes

Grand prize is a trip for two to an Open Source event anywhere in the

world

No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date:

8/11/2008

4:59 PM

--
Internal Virus Database is out-of-date.
Checked by AVG.
Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date:

8/4/2008

7:23 PM

This SF.Net email is sponsored by the Moblin Your Move Developer's

challenge

Build the coolest Linux based applications with Moblin SDK & win great

prizes

Grand prize is a trip for two to an Open Source event anywhere in the

world

No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date:

8/11/2008

4:59 PM

--
Internal Virus Database is out-of-date.
Checked by AVG.
Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date:

8/4/2008

7:23 PM

--
Internal Virus Database is out-of-date.
Checked by AVG.
Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008
7:23 PM

This SF.Net email is sponsored by the Moblin Your Move Developer's

challenge

Build the coolest Linux based applications with Moblin SDK & win great

prizes

Grand prize is a trip for two to an Open Source event anywhere in the

world

No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.138 / Virus Database: 270.6.1/1608 - Release Date: 8/12/2008

4:59 PM

--
Internal Virus Database is out-of-date.
Checked by AVG.
Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008
7:23 PM

No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.138 / Virus Database: 270.6.1/1608 - Release Date: 8/12/2008 4:59 PM

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Aug 13, 2008 6:29 AM

[OpenSIPStack] GIPs integration to OpenSIPStack

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