Thread: [OpenSIPStack] TCP and TLS support

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Hi there,

Is TCP and TLS transport protocols are supported by OpenSipStack right now? I am sure on TLS is not but still can anybody confirm this?

Thanks in advance.

Regards,

Jaydeep

> 
> Is TCP and TLS transport protocols are supported by OpenSipStack right now? I am sure on TLS is not but still can anybody confirm this?
> 

Only UDP is supported in OpenSIPStack current.  TCP support is incomplete and TLS is, of course, dependent on TCP.

Joegen

Help needed...
This is what we need to do.

We have our own RTP stack that we want to use. We just want to use the
SIP
portion of the opensipstack and have the SIP portion tell us when to set
up
an RTP call. We want to manage the RTP portion completely.

How do we do this using opensipstack?

Marshall Jobe wrote:
> Help needed...
> This is what we need to do.
>
> We have our own RTP stack that we want to use. We just want to use the
> SIP
> portion of the opensipstack and have the SIP portion tell us when to set
> up
> an RTP call. We want to manage the RTP portion completely.
>
> How do we do this using opensipstack?
>   

Please clarify.  You want to use your own RTP stack.  Does this include 
using your own codecs and audio input output steams as well?

Yes...

I want to access the SIP interface of opensipstack and let our code
handle the RTP and voice (codecs and all).

I just want to use the SIP portion. 

I have found no technical documentation whatsoever and don't know where
to go from here.

-----Original Message-----
From: jo...@op... [mailto:joe...@gm...] 
Sent: Wednesday, October 22, 2008 8:49 PM
To: ope...@li...
Subject: Re: [OpenSIPStack] TCP and TLS support

Marshall Jobe wrote:
> Help needed...
> This is what we need to do.
>
> We have our own RTP stack that we want to use. We just want to use the
> SIP
> portion of the opensipstack and have the SIP portion tell us when to
set
> up
> an RTP call. We want to manage the RTP portion completely.
>
> How do we do this using opensipstack?
>   

Please clarify.  You want to use your own RTP stack.  Does this include 
using your own codecs and audio input output steams as well?

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Documentation is scarse for opensipstack, at least as of the moment.  
Please give more info on what type of application you would want to 
build so people can point you to the right direction.  For starters, 
CallSession and CallSessionManager subclass is the way to go for 
UA/Softphone implementation.

Marshall Jobe wrote:
> Yes...
>
> I want to access the SIP interface of opensipstack and let our code
> handle the RTP and voice (codecs and all).
>
> I just want to use the SIP portion. 
>
> I have found no technical documentation whatsoever and don't know where
> to go from here.
>
>
> -----Original Message-----
> From: jo...@op... [mailto:joe...@gm...] 
> Sent: Wednesday, October 22, 2008 8:49 PM
> To: ope...@li...
> Subject: Re: [OpenSIPStack] TCP and TLS support
>
> Marshall Jobe wrote:
>   
>> Help needed...
>> This is what we need to do.
>>
>> We have our own RTP stack that we want to use. We just want to use the
>> SIP
>> portion of the opensipstack and have the SIP portion tell us when to
>>     
> set
>   
>> up
>> an RTP call. We want to manage the RTP portion completely.
>>
>> How do we do this using opensipstack?
>>   
>>     
>
> Please clarify.  You want to use your own RTP stack.  Does this include 
> using your own codecs and audio input output steams as well?
>
>
>
>
> ------------------------------------------------------------------------
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> This SF.Net email is sponsored by the Moblin Your Move Developer's
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> Build the coolest Linux based applications with Moblin SDK & win great
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> world
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> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>
> -------------------------------------------------------------------------
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> Build the coolest Linux based applications with Moblin SDK & win great prizes
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> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>   
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>   

This is what we want.

We have an RTP stack of our own that we want to use.

We need an interface to a SIP stack to initiate and receive SIP call
sessions.

We do NOT want the part of opensipstack that does the RTP.

For outgoing calls, we need something like this:

1.    Our app will setup an RTP ip/port to listen on (UDP).

2.    Our app will give this RTP ip/port information (as well as any
other necessary info) to the opensipstack
to make a call (INVITE and such).

3.    The SIP stack will negotiate with the destination SIP stack
(wherever that might be). The negotiation,
of course, will tell us what ip/port to send the RTP for outgoing voice.
We need a mechanism to receive
this info after the SIP portion of the call as been accepted by the
foreign site.

4.    Now the SIP part is done (w/o using the opensipstack RTP handling
facilities)

5.    Now it is the job of our code to send/receive RTP from the foreign
host, and walaa all it happy.

6.    We also need to be able to send/receive the call terminations
(BYEs) and such.

For incoming calls, we need something like this:

1.    We setup the SIP stack to listen on 5060.

2.    It receives incoming calls, (potentially gives this to our code
for approval).
Or maybe we just want to allow the SIP stack to handle everything SIP
relate

3.    The SIP stack will need to know what RTP port we want to listen on
(maybe it should ask us)

4.    Once a call is setup, again, it is our code that will be
sending/receiving RTP.

-----Original Message-----
From: jo...@op... [mailto:joe...@gm...] 
Sent: Thursday, October 23, 2008 10:41 AM
To: ope...@li...
Subject: Re: [OpenSIPStack] TCP and TLS support

Documentation is scarse for opensipstack, at least as of the moment.  

Please give more info on what type of application you would want to 

build so people can point you to the right direction.  For starters, 

CallSession and CallSessionManager subclass is the way to go for 

UA/Softphone implementation.

Marshall Jobe wrote:

> Yes...

> 

> I want to access the SIP interface of opensipstack and let our code

> handle the RTP and voice (codecs and all).

> 

> I just want to use the SIP portion. 

> 

> I have found no technical documentation whatsoever and don't know
where

> to go from here.

> 

> 

> -----Original Message-----

> From: jo...@op... [mailto:joe...@gm...] 

> Sent: Wednesday, October 22, 2008 8:49 PM

> To: ope...@li...

> Subject: Re: [OpenSIPStack] TCP and TLS support

> 

> Marshall Jobe wrote:

>   

>> Help needed...

>> This is what we need to do.

>> 

>> We have our own RTP stack that we want to use. We just want to use
the

>> SIP

>> portion of the opensipstack and have the SIP portion tell us when to

>>     

> set

>   

>> up

>> an RTP call. We want to manage the RTP portion completely.

>> 

>> How do we do this using opensipstack?

>>   

>>     

> 

> Please clarify.  You want to use your own RTP stack.  Does this
include 

> using your own codecs and audio input output steams as well?

> 

> 

> 

> 

>
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> 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

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> world

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> _______________________________________________

> opensipstack-devel mailing list

> ope...@li...

> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

> 

> 

> 

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

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> _______________________________________________

> opensipstack-devel mailing list

> ope...@li...

> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

>   

>
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> 

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> Version: 8.0.175 / Virus Database: 270.8.2/1740 - Release Date:
10/22/2008 7:24 PM

> 

>   

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This means windns.h was not located by configure.exe.  DNS API is 
usually part of the platform SDK that goes with visual studio.  This is 
a dependency of opensipstack in windows builds.    Please confirm if you 
have windns.h and dnsapi.lib in your system.   If not, install the 
platform SDK for you MSVC compiler

Marshall Jobe wrote:
>
>  
>
> SIPSrvRecord.cxx will not compile...
>
>  
>
> .\src\SIPSrvRecord.cxx(452) : error C2653: 'PDNS' : is not a class or 
> namespace name
>
> .\src\SIPSrvRecord.cxx(452) : error C2065: 'SRVRecordList' : 
> undeclared identifier
>
> .\src\SIPSrvRecord.cxx(452) : error C2065: 'srvUdpRecords' : 
> undeclared identifier
>
> .\src\SIPSrvRecord.cxx(453) : error C2653: 'PDNS' : is not a class or 
> namespace name
>
> .\src\SIPSrvRecord.cxx(453) : error C2065: 'SRVRecordList' : 
> undeclared identifier
>
> .\src\SIPSrvRecord.cxx(453) : error C2065: 'srvTcpRecords' : 
> undeclared identifier
>
> .\src\SIPSrvRecord.cxx(464) : error C2065: 'srvUdpRecords' : 
> undeclared identifier
>
>  
>
> ...
>
>  
>
>  
>
> I have tracked it down. The opensipstack/include/ptclib/pdns.h file is 
> #if'd out.
>
>  
>
> At the top it says #if P_DNS since this is not defined in the project, 
> it doesn't build.
>
>  
>
> Why is this not defined? I'm using your .sln file. OpenSIPStack.sln.
>
> I am using
>
>  
>
>  
>
>  
>
> -----Original Message-----
> From: jo...@op... [mailto:joe...@gm...]
> Sent: Wednesday, October 22, 2008 8:49 PM
> To: ope...@li...
> Subject: Re: [OpenSIPStack] TCP and TLS support
>
>  
>
> Marshall Jobe wrote:
>
> > Help needed...
>
> > This is what we need to do.
>
> > 
>
> > We have our own RTP stack that we want to use. We just want to use the
>
> > SIP
>
> > portion of the opensipstack and have the SIP portion tell us when to set
>
> > up
>
> > an RTP call. We want to manage the RTP portion completely.
>
> > 
>
> > How do we do this using opensipstack?
>
> >  
>
>  
>
> Please clarify.  You want to use your own RTP stack.  Does this include
>
> using your own codecs and audio input output steams as well?
>
>  
>
>  
>
>  
>
>  
>
> -------------------------------------------------------------------------
>
> 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
>
> http://moblin-contest.org/redirect.php?banner_id=100&url=/
>
> _______________________________________________
>
> opensipstack-devel mailing list
>
> ope...@li...
>
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>  
>
> ------------------------------------------------------------------------
>
>
> No virus found in this incoming message.
> Checked by AVG - http://www.avg.com 
> Version: 8.0.175 / Virus Database: 270.8.2/1740 - Release Date: 10/22/2008 7:24 PM
>
>   