[Freeswitch-users] Snom360 softphone and Freeswitch

[Brian West]

I suspect the softphone will have the same bugs the hard phone had.   
Which makes it very hard to use without a few hacks.

/b

On May 19, 2008, at 1:14 PM, Krzysiek wrote:

> 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.
>
>
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Brian West
sip:brian at freeswitch.org