Dear Kamailio users,
 
Thank you for your help about the concept of TLS & SIPS.
I still confuse with the scenerio of how to use SIPS/TLS to make security VoIP by Opensips.
I got the information like following from console after run opensips 1.4.0 with TLS enabled.
 
Listening on
             udp: 192.168.1.11 [192.168.1.11]:5060
             tcp: 192.168.1.11 [192.168.1.11]:5060
             tls: 192.168.1.11 [192.168.1.11]:5061
Aliases:
             tls: wavehost:5061
             tcp: wavehost:5060
             udp: wavehost:5060
 
As RFC3261 said, 5061 is default port for SIP. So, there are two ports here after TLS enabled. My question is how to configure my UA to use the SIP server?
In the other word, how to use SIP server with TLS enabled? My UA is SJphone 1.6, seems it doesn't support SIPs/TLS. Is there any open source SIP UA which support TLS availiable ?
 
Best regards,
Steven Wu

From: users-bounces@lists.kamailio.org 代表 Henning Westerholt
Sent: 2009-2-25 (星期三) 18:05
To: users@lists.kamailio.org
Cc: Steven Wu
Subject: Re: [Kamailio-Users] Secure VoIP

On Wednesday 25 February 2009, Steven Wu wrote:
> Does anybody know how to configure OpenSER 1.3 to support secure VoIP?
>
> Another question about SIP is how difference between sips and TLS?

Hi Steven,

you find some documentation about TLS support in kamailio/ OpenSER here:
http://kamailio.org/docs/tls.html.

SIPS is the secure variant of SIP, it uses TLS to encrypt its data.

Cheers,

Henning

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