Chris, The a=encryption:optional tag is a Microsoft extension held over (Snom has removed it in recent beta builds I have). We do not support it and will not. Grandstream uses RTP/SAVP, Snom will be doing RTP/ SAVP in 7.1.33 (in beta), Polycom does the dual m= line for RTP/SAVP and RTP/AVP. Offering crypto in an RTP/AVP is wrong please refer to RFC3711 where it clearly states "The RTP specification establishes a registry of profile names for use by higher-level control protocols, such as the Session Description Protocol (SDP), to refer to transport methods. This profile registers the name "RTP/SAVP"." If he wishes to see the most recent Snom beta where they are now doing it correctly please have him contact me. /b PS RFC3711 section 12 On Apr 24, 2008, at 4:35 AM, Krzysiek wrote:
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@172.16.115.114 SIP/2.0 Via: SIP/2.0/UDP 172.16.115.114 :5070;branch=z9hG4bK0014de144110dd118b3d00123fa91863;rport From: <sip:hs@172.16.115.114>;tag=3538012975 To: <sip:123@172.16.115.114> Call-ID: 0014DE14-4110-DD11-8B3C-00123FA91863@172.16.115.114 CSeq: 1 INVITE Contact: <sip:hs@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@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@freeswitch.org
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Brian West sip:brian@freeswitch.org