[Kamailio-Users] about kamailio and tls

Daniel-Constantin Mierla miconda at gmail.com
Mon Jan 11 11:06:39 CET 2010


Hi,

On 1/9/10 11:01 PM, Klaus Darilion wrote:
> Hi Mustafa!
>
> mustafa samara wrote:
>>  i am mustafa samara master degree student. i try to test qjsimple 
>> with kamailio openser is it possible (to test the tls support) ?
>
> Yes, this should be no problem. Just configure TLS support in kamailio 
> (I suggest to use kamailio 1.5). 
I think today's kamailio 3.0.0 is also good canditate. TLS is easier to 
load, being a module and has several improvements.

Cheers,
Daniel

> For testing with pjsip you can either use pjsua client (included in 
> pjsip), or you could also use QjSimple 
> (http://www.ipcom.at/index.php?id=560) which is a prototype SIP client 
> based on pjsip with support for TLS and SRTP.
>
>> also i want to ask about ( in sip preferences) what is the deference 
>> when we you use tls as a protocol or when we use (tls or sips) as a 
>> SRTP requirements.
>
> First you have to differ between signaling and media transport. 
> Signaling uses SIP, media transport uses (at least for audio and 
> video) RTP. For both protocols exists mechanisms to encrypt the payload.
>
> If you want to encrypt SIP, you can use SIP over TLS, thus the SIP 
> signaling is encrypted.
>
> If you want to encrypt the media transport you use SRTP. With SRTP 
> only the media payload itself is encrypted.
>
> There are a few differences between SIP and RTP encryption. When using 
> SIP over TLS - the whole SIP signaling is encrypted - but only between 
> the hops which use TLS as transport. For example if a client (caller) 
> sends the SIP message with UDP to the proxy, and the proxy forwards 
> the SIP message to another client (callee) over TLS, only the part 
> between proxy and the callee is encrypted.
>
> When using SRTP, not the whole message, but only the media payload is 
> encrypted. The RTP headers are still sent in clear text. Usually the 
> encryption is end-to-end between caller and callee.
>
> Now, as you see, SIP and RTP a rather independent. You can use SIP 
> over TLS and RTP, you can use SIP over UDP and SRTP, or you can use 
> SIP over TLS and SRTP. Thus, from a technical point of view you can 
> encrypt signaling, media, or both.
>
> For SRTP, both parties need to know a shared secret - the encryption 
> key. There are several methods for SRTP key exchange (google for: srtp 
> sdes mikey dtls). Currently the most used SRTP key exchange is "SDES" 
> (RFC 4568). With SDES, the encryption key is exchanged in the session 
> description (SDP) - similar to codec negotiation.
>
> When using SDES, the encryption is in plain text in the SDP. Thus, 
> sending SIP over unencrypted transports but using SRTP is rather 
> nonsense, as the attacker can get the key from the unencrypted SDP and 
> decrypt the SRTP packet.
>
> Thus, when using SDES, some SIP clients (e.g. pjsip) give you the 
> configuration option to use SRTP (with SDES) only if the SIP signaling 
> is sent over encrypted transport (TLS).
>
> Finally the difference between TLS and SIPS: TLS can be used as 
> transport (just like UDP or TCP) between any hops. When addressing a 
> target with a sip: URI, the SIP nodes can use any of these protocols 
> the send the SIP message. When addressing a target with a sips: URI, 
> the standard requires that the message is sent from sender to receiver 
> over encrypted transport. As a practical result: A message to a sip: 
> URI can use any transport (UDP,TCP,TLS) whereas a mesage to a sips: 
> URI must use encrypted transport on every hop (TLS).
>
> regards
> Klaus
>
>  - the RTP header is still in plain text (this is different to
>
>>
>> clould you help me please?
>> i wat to include the result in my thesis
>>
>> -- 
>> Eng.Mustafa Al-Samara
>>
>>
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>>
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>
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-- 
Daniel-Constantin Mierla
* http://www.asipto.com/





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