[Freeswitch-users] SRTP in PhonerLite and Freeswitch

[Brian West]

Chris,
	Heiko and I have had a few exchanges in email... He's a very nice guy  
very preceptive to the changes required to support this properly.  He  
said he'll be trying out the two m-line method soon.

/b

On Apr 24, 2008, at 4:35 AM, Krzysiek wrote:

> I have just received replay from the author of PhonerLite:
>
> "Hi Chris,
> I don't think it is a bug. I do it like others do too.
> For PhonerLite SRTP is optional (see: a=encryption:optional). So if  
> Freeswitch doesn't support this, SRTP fails and RTP is unencrypted.
> This is exactly the the way like Snom and Grandstream offer SRTP in  
> their SDP.
>
> I won't change this!
>
> Heiko
>
>
> INVITE sip:123 at 172.16.115.114 SIP/2.0
> Via: SIP/2.0/UDP  
> 172.16.115.114 
> :5070;branch=z9hG4bK0014de144110dd118b3d00123fa91863;rport
> From: <sip:hs at 172.16.115.114>;tag=3538012975
> To: <sip:123 at 172.16.115.114>
> Call-ID: 0014DE14-4110-DD11-8B3C-00123FA91863 at 172.16.115.114
> CSeq: 1 INVITE
> Contact: <sip:hs at 172.16.115.114:5070>
> Content-Type: application/sdp
> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE,  
> UPDATE
> Max-Forwards: 70
> Supported: 100rel, replaces
> User-Agent: SIPPER for PhonerLite
> Content-Length:   506
>
> v=0
> o=- 2350732611 0 IN IP4 172.16.115.114
> s=SIPPER for PhonerLite
> c=IN IP4 172.16.115.114
> t=0 0
> m=audio 5072 RTP/AVP 0 8 2 3 97 110 111 9 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 iLBC/8000
> a=rtpmap:110 speex/8000
> a=rtpmap:111 speex/16000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:PYoCXAWH4BjD/tmuK+Ww5/ 
> pIa9MIBT424Judwmty
> a=encryption:optional
> a=fmtp:101 0-15
> a=sendrecv "
>
> :(
> Chris
> ----- Original Message -----
> From: Brian West
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thursday, April 24, 2008 1:17 AM
> Subject: Re: [Freeswitch-users] SRTP in PhonerLite and Freeswitch
>
> Well the snom on 7.1.33 will be correct... the Grandstream will work  
> correctly also.
>
> /b
>
> On Apr 23, 2008, at 5:31 PM, Krzysiek wrote:
>> Thanks for help and explanation.
>> Yes I know that I should use TLS. This was just a test.
>> That's a pity, that this client is broken. I have just thought that  
>> I found nice and free sip client with srtp and tls support. I have  
>> found this client here: http://www.phonerlite.de/ , author Heiko  
>> Sommerfeldt. I hope, he corrects this bug.
>>
>> >The Polycom is the only phone that does this little tid bit  
>> correctly.
>>
>> Do I understand this correctly that there aren't any softphone  
>> clients that support SRTP via SDES correctly ?  I found few clients  
>> that support SRTP here http://en.wikipedia.org/wiki/Comparison_of_VoIP_software 
>>  but  I haven't tested them yet.
>
> Brian West
> sip:brian at freeswitch.org
>
>
>
>
>
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Brian West
sip:brian at freeswitch.org



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