[Freeswitch-users] SRTP in PhonerLite and Freeswitch

[Michael Jerris]

Can you post a sip trace of this entire call, the 19 means we are  
rejecting that m= line, are there 2 m lines, AVP and SAVP to indicate  
optional secure?

Mike

On Apr 23, 2008, at 3:01 PM, Krzysiek wrote:

> Hi
> I have 2 softphones PhonerLite (they support SRTP via SDES ) and the  
> freeswitch (windows RC1 version) server and I wanted to make secure  
> call between those two endpoints (SRTP).
> I spend whole day on testing this scenario and my conclusions are:
> - when the option: <action application="export"  
> data="sip_secure_media=true"/> is uncommented, and both enpoints  
> have enabled SRTP then:
> 1) Initiator of the session sends SIP Invite with a=crypto paramter  
> and supported codecs
> 2) Freeswitch receives SIP Invite and sends SIP Invite to the  
> receiver (also with the crypto)
> 3) Receiver receives the SIP Invite with the a=crypto parameter and  
> he sends back supported codecs with 200 OK message (but without  
> a=crypto parametr. Is that ok? I'm afraid not)
> 4) Freeswitch sends 200 OK message but witout any codecs: m=audio 0  
> RTP/AVP 19 and no a= parameters!
> 5) Final result is that the second leg of the session between  
> Freeswitch and receiver has SRTP transport enbaled and the first leg  
> (initiator- Freeswitch) doesn't hear anything - no codecs! However  
> Freeswitch is sending RTP (not SRTP) pacekets to the initiator.
>
> Could someone explain to me, what is going on, and why freeswitch  
> doesn't forward codecs accepted by the receiver to the initiator?
> Is it a PhonerLite's bug or freeswitch's? Maybe someone has tested  
> SRTP with the PhonerLite softphone or any other free softphone with  
> srtp support?
>
> When I uncommented: <param name="Inbound-no-media" value="true">
> everything works fine. The parameter <action application="export"  
> data="sip_secure_media=true"/> doesn't change anything then (but i  
> cound miss something).
>
> Thanks for help
> Chris
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