[Freeswitch-users] SRTP in PhonerLite and Freeswitch

Krzysiek cris7 at o2.pl
Thu Apr 24 02:35:24 PDT 2008


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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