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:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/TLS 192.168.1.4:1145;branch=z9hG4bK-k1yp4ytettb1;rport > From: "1001" <sip:[EMAIL PROTECTED]>;tag=9udth2g3o3 > To: <sip:[EMAIL PROTECTED];user=phone> > Call-ID: [EMAIL PROTECTED] > CSeq: 1 INVITE > Max-Forwards: 70 > Contact: > <sip:[EMAIL PROTECTED]: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. > > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West sip:[EMAIL PROTECTED] _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
