[Freeswitch-users] Snom360 softphone and Freeswitch

[Krzysiek]

Hi

I have recently tested snom 360 softphone 
(www.snom.com/download/snom360-5.3.exe) with Freeswitch. It has SRTP* and 
TLS support. It is quite old (2006) and I'm wondering if this softphone 
creates sip sessions (with tls enabled) properly.
Here is a sip trace (INVITE message should be sufficient) :

 ------------------------------------------------------------------------
recv 1092 bytes from tls/[192.168.1.4]:1145 at 17:09:19.945705:
   ------------------------------------------------------------------------
   INVITE sip:1002 at 192.168.1.2;user=phone SIP/2.0
   Via: SIP/2.0/TLS 192.168.1.4:1145;branch=z9hG4bK-k1yp4ytettb1;rport
   From: "1001" <sip:1001 at 192.168.1.2>;tag=9udth2g3o3
   To: <sip:1002 at 192.168.1.2;user=phone>
   Call-ID: 50b43148ee7f-696c59fq7vjl at snomSoft-000413FFFFFF
   CSeq: 1 INVITE
   Max-Forwards: 70
   Contact: 
<sip:1001 at 192.168.1.4:1145;transport=tls;line=ojn9itpa>;flow-id=1
   P-Key-Flags: resolution="31x13", keys="4"
   User-Agent: snomSoft/5.3
   Accept: application/sdp
   Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, 
PRACK, MESSAGE, INFO
   Allow-Events: talk, hold, refer
   Supported: timer, 100rel, replaces, callerid
   Session-Expires: 3600;refresher=uas
   Content-Type: application/sdp
   Content-Length: 362

   v=0
   o=root 28625 28625 IN IP4 192.168.1.4
   s=call
   c=IN IP4 192.168.1.4
   t=0 0
   m=audio 57428 RTP/AVP 0 8 3 101
   a=crypto:1 AES_CM_128_HMAC_SHA1_32 
inline:IZpDi51+JeVKPOO3Mox0q3jZYmJorsThpl6b2jw1
   a=rtpmap:0 pcmu/8000
   a=rtpmap:8 pcma/8000
   a=rtpmap:3 gsm/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=encryption:optional
   a=sendrecv

I have read in rfc3261 that :
"The use of "transport=tls" has consequently been deprecated"

So does snom 360 create sessions properly or not? Freeswitch works with it 
quite good but I don't know if it is right according to rfc.
I tested few free/opensource softphones such as Minisip, Lynxphone, 
PhonerLite and neither worked properly with freeswitch with srtp/tls 
enabled. PhonerLite crashed when the call was to be setup although it 
registered properly , Minisip created tls sip sessions in weird way 
(transport=tcp) that the second leg of the call wasn't created (I wrote 
about this here 
http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003179.html), 
and Lynxphone didn't even register.

So snom will be first FREE softphone that works.

*I know that this softphone has bad SRTP/SDES implementation (RTP/AVP 
instead of RTP/SAVP) but I think that it isn't such a big problem when media 
streams by-pass Freeswitch server (for example when calls are setup in local 
LAN and Inbound-no-media is set to true). Am I right?

TIA for reply
Chris

PS Sorry for my english.