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<DIV><FONT face=Arial size=2><FONT face="Times New Roman" size=3>I have just
received replay from the author of PhonerLite:</FONT> </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>"</FONT>Hi Chris,<BR></DIV>
<DIV>I don't think it is a bug. I do it like others do too.<BR>For PhonerLite
SRTP is optional (see: a=encryption:optional). So if Freeswitch doesn't support
this, SRTP fails and RTP is unencrypted.<BR>This is exactly the the way like
Snom and Grandstream offer SRTP in their SDP.<BR><BR>I won't change
this!<BR><BR>Heiko<BR><BR><BR>INVITE <A class=moz-txt-link-freetext
href="sip:123@172.16.115.114">sip:123@172.16.115.114</A> SIP/2.0<BR>Via:
SIP/2.0/UDP
172.16.115.114:5070;branch=z9hG4bK0014de144110dd118b3d00123fa91863;rport<BR>From:
<A class=moz-txt-link-rfc2396E
href="sip:hs@172.16.115.114"><sip:hs@172.16.115.114></A>;tag=3538012975<BR>To:
<A class=moz-txt-link-rfc2396E
href="sip:123@172.16.115.114"><sip:123@172.16.115.114></A><BR>Call-ID: <A
class=moz-txt-link-abbreviated
href="mailto:0014DE14-4110-DD11-8B3C-00123FA91863@172.16.115.114">0014DE14-4110-DD11-8B3C-00123FA91863@172.16.115.114</A><BR>CSeq:
1 INVITE<BR>Contact: <A class=moz-txt-link-rfc2396E
href="sip:hs@172.16.115.114:5070"><sip:hs@172.16.115.114:5070></A><BR>Content-Type:
application/sdp<BR>Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY,
MESSAGE, UPDATE<BR>Max-Forwards: 70<BR>Supported: 100rel,
replaces<BR>User-Agent: SIPPER for PhonerLite<BR>Content-Length:
506<BR><BR>v=0<BR>o=- 2350732611 0 IN IP4 172.16.115.114<BR>s=SIPPER for
PhonerLite<BR>c=IN IP4 172.16.115.114<BR>t=0 0<BR>m=audio 5072 RTP/AVP 0 8 2 3
97 110 111 9 101<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:2
G726-32/8000<BR>a=rtpmap:3 GSM/8000<BR>a=rtpmap:97 iLBC/8000<BR>a=rtpmap:110
speex/8000<BR>a=rtpmap:111 speex/16000<BR>a=rtpmap:9 G722/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:PYoCXAWH4BjD/tmuK+Ww5/pIa9MIBT424Judwmty<BR>a=encryption:optional<BR>a=fmtp:101
0-15<BR>a=sendrecv "</DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>:(</FONT></DIV>
<DIV><FONT face=Arial size=2>Chris</FONT></DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=brian@freeswitch.org href="mailto:brian@freeswitch.org">Brian
West</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, April 24, 2008 1:17
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] SRTP in
PhonerLite and Freeswitch</DIV>
<DIV><BR></DIV>Well the snom on 7.1.33 will be correct... the Grandstream will
work correctly also.
<DIV><BR></DIV>
<DIV>/b</DIV>
<DIV><BR>
<DIV>On Apr 23, 2008, at 5:31 PM, Krzysiek wrote:<BR
class=Apple-interchange-newline>
<BLOCKQUOTE type="cite"><SPAN class=Apple-style-span
style="WORD-SPACING: 0px; FONT: 12px Helvetica; TEXT-TRANSFORM: none; COLOR: rgb(0,0,0); TEXT-INDENT: 0px; WHITE-SPACE: normal; LETTER-SPACING: normal; BORDER-COLLAPSE: separate; orphans: 2; widows: 2; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0">
<DIV><FONT face=Arial size=2>Thanks for help and
explanation. </FONT></DIV>
<DIV><FONT face=Arial size=2>Yes I know that I should use TLS.
This was just a test. </FONT></DIV>
<DIV><FONT face=Arial size=2>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:<SPAN
class=Apple-converted-space> </SPAN></FONT><FONT face=Arial size=2><A
href="http://www.phonerlite.de/">http://www.phonerlite.de/</A> , author
Heiko Sommerfeldt. I hope, he corrects this bug.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>><FONT face="Times New Roman" size=3>The
Polycom is the only phone that does this little tid bit
correctly.</FONT></FONT></DIV>
<DIV><FONT face=Arial size=2><FONT face=Arial
size=2></FONT></FONT> </DIV>
<DIV><FONT face=Arial size=2>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<SPAN
class=Apple-converted-space> </SPAN><A
href="http://en.wikipedia.org/wiki/Comparison_of_VoIP_software">http://en.wikipedia.org/wiki/Comparison_of_VoIP_software</A> but<SPAN
class=Apple-converted-space> </SPAN> I haven't tested them
yet.</FONT></DIV></SPAN></BLOCKQUOTE></DIV><BR>
<DIV apple-content-edited="true"><SPAN class=Apple-style-span
style="WORD-SPACING: 0px; FONT: 12px Helvetica; TEXT-TRANSFORM: none; COLOR: rgb(0,0,0); TEXT-INDENT: 0px; WHITE-SPACE: normal; LETTER-SPACING: normal; BORDER-COLLAPSE: separate; orphans: 2; widows: 2; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0">
<DIV
style="WORD-WRAP: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space">
<DIV>Brian West</DIV>
<DIV><A href="sip:brian@freeswitch.org">sip:brian@freeswitch.org</A></DIV>
<DIV><BR></DIV></DIV></SPAN><BR
class=Apple-interchange-newline></DIV><BR></DIV>
<P>
<HR>
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