Snom360 softphone and Freeswitch
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@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@192.168.1.2>;tag=9udth2g3o3 To: <sip:1002@192.168.1.2;user=phone> Call-ID: 50b43148ee7f-696c59fq7vjl@snomSoft-000413FFFFFF CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:1001@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.
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@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@192.168.1.2>;tag=9udth2g3o3 To: <sip:1002@192.168.1.2;user=phone> Call-ID: 50b43148ee7f-696c59fq7vjl@snomSoft-000413FFFFFF CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:1001@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.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West sip:brian@freeswitch.org
participants (2)
-
Brian West
-
Krzysiek