[Freeswitch-users] SIP Contact Header Issue When Using TLS

Michael Jerris mike at jerris.com
Tue Mar 25 20:22:58 MSK 2014


sips: should not make a difference, however.. take a look at bind-params and tls-bind-params

https://wiki.freeswitch.org/wiki/Sofia.conf.xml

On Mar 25, 2014, at 1:15 PM, JP <jaykris at gmail.com> wrote:

> Is there any way to specify the full contact header in a UA profile that the SIP stack will use when formulating messages?  Specifically, have it use "sips" instead of "sip" as the protocol scheme? 
> 
>  
> I'm trying to establish an INVITE dialog between 2 FreeSWITCH servers using a client authenticated TLS handshake.
> 
>  
> To accomplish this, I've created 2 UA profiles on both servers - one to fulfill the role of the UAC (i.e. tls-uac.xml) and one to implement the UAS (i.e. tls-uas.xml).  Here are the relevant parameters from both profiles:
> 
>  
> tls-uac.xml:
> 
>  
>             <param name="sip-port" value="5081"/>
> 
>  
>             <param name="tls" value="true"/>
> 
>  
>             <param name="tls-only" value="true"/>
> 
>  
>             <param name="tls-sip-port" value="5082"/>
> 
>  
>             <param name="tls-cert-dir" value="$${base_dir}/conf/tls/uac"/>
> 
>  
> tls-uas.xml:
> 
>  
>             <param name="sip-port" value="5081"/>
> 
>  
>             <param name="tls" value="true"/>
> 
>  
>             <param name="tls-only" value="true"/>
> 
>  
>             <param name="tls-sip-port" value="5081"/>
> 
>  
>             <param name="tls-cert-dir" value="$${base_dir}/conf/tls/uas"/>
> 
>  
> The problem already starts when "tls-uac" sends a non-secure SIP URI in the contact header of its initial INVITE request (i.e. sip:mod_sofia at 10.191.210.150:5081).  But the more immediate issue is that "tls-uas" also responds with a non-secure SIP URI in the contact header of its final response (i.e. sip:14086805675 at 10.191.210.151:5081;transport=udp).  This causes "tls-uac" to send its ACK to the right port number (i.e. 5081) but on the wrong transport (i.e. UDP instead of TCP/TLS).
> 
>  
> I've seen in the FS documentation that there are ways to manipulate the contact header through the dial plan, but I'd really prefer not to do it this way.  Any suggestions?
> 
>  
> Thanks
> 
> JP
> 
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