[asterisk-dev] Testing of SIP TCP/TLS support

[Paul Bryson]

Does anyone know the status of this branch?  I'm very interested in
the SIP over TCP support and was wondering where it is at.  Will this
be merged into a 1.4.x branch soon?  Has community testing gone well?
Is there a post or site that provides this information?  I haven't
been able to locate anything.


Paul Bryson

On 07/09/07, Russell Bryant <russell at digium.com>  wrote:
> Brett Bryant has been working as an intern here at Digium for the
> Summer.  After completing his updates to Mantis, he has been working on
> various Asterisk bugs and new features.  One of these has been working
> on TCP and TLS support for SIP.  I think that there is still some work
> to be done to get the client side of TLS working, but it's close.  The
> TCP part appears to be working well in our tests here.  Also, accepting
> TLS connections should be working, as well.  I'd like to invite anyone
> interested to take a look and test it out.
>
> $ svn co http://svn.digium.com/svn/asterisk/team/bbryant/sip-tcptls
>
> Please send any feedback to this mailing list.
>
>
> Here is a quick reference to the configuration changes.  All of these
> are in configs/sip.conf.sample, as well.
>
> To enable listening for TCP connections, there are 3 options in the
> general section: tcpenable, tcpbindaddr, and tcpbindport.
>
> To enable listening for TLS connections, there are 4 options in the
> general section: tlsenable, tlsbindaddr, tlsbindport, and tlscertfile.
>
> To specify a transport to be used for a registration, it is specified at
> the beginning of the register statement.  For example:
>
> register => tcp://russell:password@digium.com/1234
>
> To specify a transport to use when connecting to a peer, you can put the
> "transport" option in a peer section.  For example, "transport=tcp".
>
> --
> Russell Bryant
> Software Engineer
> Digium, Inc.
>