Open Source Sip: [OpenSIPStack] is it necessary to parse ...

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24 Replies Last post: Aug 15, 2007 6:30 AM by Guest Go to original post 1 2 Previous Next

Guest

15. Aug 14, 2007 1:21 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib

Hello Whit,

No such thing as a dumb question.

Whit Thiele wrote:

This is a question related to the problems I''ve been having with the
ntdll.dll crash. I just want to ensure that I am using the correct libraries
and installing everything correctly.

When I compile ATLSIP in Release, it generates the ATLSIP.dll

When I compile ATLSIP in Release-Minimal mode, it generates the
AxInterop.ATLSIPLib.1.0.dll file in the Release folder, hover its timestamp
is when I first downloaded ATLSIP out of CVS, so I''m assuming this isn''t
compiled, but copied.

It is compiled but not always. The Interop DLLs are there for
interoperability between .NET and Win32/ATL. So if there are no major
structural changes to ATLSIP, they don''t need to compiled. Just check if
your AxInterop.ATLSIPLib.1.0.dll and Interop.ATLSIPLib.1.0.dll of your
Release and Interop folders have the same timestamps.

Also, one quirk I found with VS is that it doesn''t always automatically
copy Interop.ATLSIPLib.1.0.dll to the target folder. You may have to
manually copy this to your Release folder.

Which library should be used in a .NET app as a reference? Should you import
the ATLSIP.dll directly, or should you use the Interop library?

You should only add ATLSIP as a reference. VS will take care of the rest.

Ilian

Hopefully there is no such thing as a ''dumb'' question!

Regards,

Whit

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Guest

16. Aug 14, 2007 5:44 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib

Hello Ilian,
I have sent the sip.log file, have you find something?
I''m still having crashes.

Date: Tue, 14 Aug 2007 13:21:10 +0800
From: ipinzon@solegysystems.com
To: opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack ATLSIPLib and Interop.AxATLSIPLib

Hello Whit,

No such thing as a dumb question.

Whit Thiele wrote:

This is a question related to the problems I''ve been having with the
ntdll.dll crash. I just want to ensure that I am using the correct libraries
and installing everything correctly.

Which library should be used in a .NET app as a reference? Should you import
the ATLSIP.dll directly, or should you use the Interop library?

You should only add ATLSIP as a reference. VS will take care of the rest.

Ilian

Hopefully there is no such thing as a ''dumb'' question!

Regards,

Whit

_________________________________________________________________
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Guest

17. Aug 14, 2007 8:03 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib

Hi Yacine,

Yacine Auczone wrote:

Hello Ilian,
I have sent the sip.log file, have you find something?
I''m still having crashes.

I can''t tell directly from your logs what''s causing your problem. And I
still can''t reproduce this, unfortunately.

Can you do the following again?

1. Make sure that your OpenSIPStack code is up-to-date
2. Rebuild ATLSIP (clean all dlls first; atlsip.dll, *interop dlls)
3. Rebuild OSSPhone (you may have to manually add references to
AxInterop and Interop dlls)

Also, may I know a few more details how to bug occurs? Like how many
calls before the crash, the length of the calls, the settings you set in
OSSPhone, etc. Does this occur also in Debug build? If so, can your
debugger attach to it and see where the crash comes from?

Whit,

I take back what I said earlier: "You should only add ATLSIP as a
reference. VS will take care of the rest." For some reason, VS
sometimes doesn''t automatically load the AxInterop dll. You may have
to manually add this. Make sure all the references to the *Interop dlls
are up-to-date.

Regards,
Ilian

Guest

18. Aug 14, 2007 9:39 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib

Ilian,

Thanks for the suggestions. I''m going to rebuild ATLSIP and test it out
today. I''ll post how it goes or what I discover.

By the way, I''ve noticed that there is a TON of useful information in this
forum. Is there any plan to consolidate this into an online wiki or
something so that people could post tutorials, setups, experiences etc.
Things which you wouldn''t normally see or post in an email forum, but
provides an invaluable resource to end users.

Food for thought..

Whit

-----Original Message-----
From: opensipstack-devel-bounces@lists.sourceforge.net
opensipstack-devel-bounces@lists.sourceforge.net On Behalf Of Ilian
Jeri C. Pinzon
Sent: Tuesday, August 14, 2007 7:03 AM
To: opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack ATLSIPLib and Interop.AxATLSIPLib

Hi Yacine,

Yacine Auczone wrote:

Hello Ilian,
I have sent the sip.log file, have you find something?
I''m still having crashes.

Guest

19. Aug 14, 2007 1:34 PM

in response to: Guest

Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib

Ilian,

Where are the Interop dll''s created? I''ve tried looking through the project
settings but I can''t see where they are being created.

It seems (from the datestamps) that the Interop files are being copied from
somewhere, not created.

Whit

Whit,

I take back what I said earlier: "You should only add ATLSIP as a
reference. VS will take care of the rest." For some reason, VS
sometimes doesn''t automatically load the AxInterop dll. You may have
to manually add this. Make sure all the references to the *Interop dlls
are up-to-date.

Regards,
Ilian

Guest

20. Aug 14, 2007 2:07 PM

in response to: Guest

Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib

I think I''m on the right track generating the Interop assemblies here but
please correct me if I''m wrong:

References:

"COM Interop Part 1: C# Client Tutorial"

http://msdn2.microsoft.com/en-us/library/aa645736(vs.71).aspx#vcwlkcomintero
ppart1cclienttutorialanchor1

"Type Library Importer (Tlbimp.exe)"

http://msdn2.microsoft.com/en-us/library/tt0cf3sx(VS.80).aspx

"Generating Primary Interop Assemblies"

http://msdn2.microsoft.com/en-us/library/dwe56e27(VS.80).aspx

So I simply ran this command:

C:\NEWSIP\atlsip\Release>tlbimp ATLSIP.dll /out:MyATLSIP.dll

Subsequently running:

C:\NEWSIP\atlsip\Release>Ildasm MyATLSIP.dll

Will show the content (definitions) of this new Interop library.

This process creates a new Interop assembly for ATLSIP.dll which can then be
used in the project. So by my reasoning, this is how the original
Interop.AxATLSIPLib files were/are created.

I''m very new to .NET so this assembly stuff is a bit challenging to get my
head around.

If anyone sees an error in this approach let me know, otherwise perhaps it
will help others! Look forward to feedback!

Regards,

Whit

-----Original Message-----
From: opensipstack-devel-bounces@lists.sourceforge.net
opensipstack-devel-bounces@lists.sourceforge.net On Behalf Of Whit
Thiele
Sent: Tuesday, August 14, 2007 12:35 PM
To: opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack ATLSIPLib and Interop.AxATLSIPLib

Guest

21. Aug 14, 2007 3:22 PM

in response to: Guest

Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib

I''m sure that there is something wrong since the compilation process works great and also the app works
I think that it should work on any x86 windows xp PC but i still have troubles on many computers.
did the developpers tested any ATLSIP.dll based softphone on other computer thene the workstation?
why the ossphone installer still not updated?
ossphone installer that uses an non updated version of ATLSIP.dll works great on any PC.

Date: Tue, 14 Aug 2007 13:07:34 -0500
From: dev@whit.ca
To: opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack ATLSIPLib and Interop.AxATLSIPLib

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Guest

22. Aug 14, 2007 3:46 PM

in response to: Guest

Re: [OpenSIPStack] is it necessary to parse RTP in openSBC

Hi Joegen,
Please reply for my below mail.

Thanks for the reply.
Is there any design document about openSBC, which will tell me in detail
about how it is implementing the NAtting/ ALG functinality and how it will
handle the Media streams.
For Call Transfer, we will use the relay approach.
Also, if i want to just relay the Media packets, can you let me know the
algorithm you have applied in the openSBC.

Also, in openSBC product, is High Availability supported or it is in roadmap
?

On 8/10/07, Ashish Khare wrote:

Hi Baclor,

Thanks for the reply.
Is there any design document about openSBC, which will tell me in detail
about how it is implementing the NAtting/ ALG functinality and how it will
handle the Media streams.
For Call Transfer, we will use the relay approach.
Also, if i want to just relay the Media packets, can you let me know the
algorithm you have applied in the openSBC.

Also, in openSBC product, is High Availability supported or it is in
roadmap ?

On 8/10/07, Joegen E. Baclor wrote:

inline...

Ashish Khare wrote:

Hi Baclor,
This is still not clear to me.
Lets take a example:
Sip Client A is talking to Sip Client B through Proxy/B2BUA(P) which
handles only SIP signaling messages.
Now in Call, Sip Client A is trasnferred to Sip CLient C and now B and
C are talking.
But still they are abel to talk.
Then how this case is different from yours. Can you please elaborate
and explain to me.

There are two ways OpenSBC handles REFER. The default is to relay the
REFER to the UA and let the UA do the transfer request. This is ok
because the UA knows that there will be a change in the audio session.
The second way (Local REFER) will not relay the REFER. Instead OpenSBC
do the transfer. This leaves the other UA to not know that the call is
actually transfered. If the transfer succeeded, a new media would with

a different SSRC would have been created. If OpenSBC just relays that,
the UA may reject the packets because the ssrc has already changed.

we are considering to build ALG. We have our own SIP stack ( Proxy and

B2BUA ) but we dont have RTP stack. Also we dont want to parse the
RTP stream. Just Rely it, based on source and destination IP and
ports. Is this feasible ?. We are also exploring your openSBC if we
can used it.

Of course this is feasible. You will have to change some lines of code
in the media interface but it wont take much. Just post questions
about the code if you need to clarify something.

Guest

23. Aug 14, 2007 11:12 PM

in response to: Guest

Re: [OpenSIPStack] is it necessary to parse RTP in openSBC

Ashish Khare wrote:

Hi Joegen,
Please reply for my below mail.

Thanks for the reply.
Is there any design document about openSBC, which will tell me in detail
about how it is implementing the NAtting/ ALG functinality and how it will
handle the Media streams.

No there isn''t yet. Your best bet is to analyze the code and ask
questions.

For Call Transfer, we will use the relay approach.

Then you may not need to parse RTP

Also, if i want to just relay the Media packets, can you let me know the
algorithm you have applied in the openSBC.

RTP is parsed in RTPSession::OnReceiveData(). You may override this
function to not parse anything and just return e_ProcessPacket for all
cases. I haven''t done this myself so i''m not sure what side effects
there will be. If that happens, you are on your own.

Also, in openSBC product, is High Availability supported or it is in
roadmap
?

It is planned but did not get a date yet in the roadmap.

On 8/10/07, Ashish Khare > wrote:

Hi Baclor,

Thanks for the reply.
Is there any design document about openSBC, which will tell me in
detail about how it is implementing the NAtting/ ALG functinality
and how it will handle the Media streams.
For Call Transfer, we will use the relay approach.
Also, if i want to just relay the Media packets, can you let me
know the algorithm you have applied in the openSBC.

Also, in openSBC product, is High Availability supported or it is
in roadmap ?

On 8/10/07, Joegen E. Baclor > wrote:

inline...

Ashish Khare wrote:

Hi Baclor,
This is still not clear to me.
Lets take a example:
Sip Client A is talking to Sip Client B through

Proxy/B2BUA(P) which

handles only SIP signaling messages.
Now in Call, Sip Client A is trasnferred to Sip CLient C and

now B and

C are talking.
But still they are abel to talk.
Then how this case is different from yours. Can you please

elaborate

and explain to me.

There are two ways OpenSBC handles REFER. The default is to
relay the
REFER to the UA and let the UA do the transfer request. This
is ok
because the UA knows that there will be a change in the audio
session.
The second way (Local REFER) will not relay the REFER.
Instead OpenSBC
do the transfer. This leaves the other UA to not know that
the call is
actually transfered. If the transfer succeeded, a new media
would with
a different SSRC would have been created. If OpenSBC just
relays that,
the UA may reject the packets because the ssrc has already
changed.

we are considering to build ALG. We have our own SIP stack (

Proxy and

B2BUA ) but we dont have RTP stack. Also we dont want to

parse the

RTP stream. Just Rely it, based on source and destination IP and
ports. Is this feasible ?. We are also exploring your openSBC

if we

can used it.

Of course this is feasible. You will have to change some
lines of code
in the media interface but it wont take much. Just post
questions
about the code if you need to clarify something.

Guest

24. Aug 15, 2007 6:30 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib

Hey guys,

I''ve made several changes on OSSPhone.

You should now be able to build the OSSPhone project directly in its own
solution file without first building ATLSIP separately. The solution
file is on the OSSPhone folder.

I''ve modified this to build and reference the interop dlls automatically
before OSSPhone''s build event. I''ve also added a much cleaner exit
behavior. Also make sure you have the latest OpenSIPStack code.

Get this from CVS. Update me if you have problems.

Ilian

Yacine Auczone wrote:

Date: Tue, 14 Aug 2007 13:07:34 -0500
From: dev@whit.ca
To: opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack ATLSIPLib and Interop.AxATLSIPLib

_________________________________________________________________
Exprimez-vous : créez la page d''accueil qui vous ressemble avec Live.com.
http://www.live.com/getstarted

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