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<DIV><FONT face=Arial size=2>Sorry for such a long post here :). I was using
wireshark and it looks like this (the 4 most important messages)
:</FONT><FONT face=Arial
size=2><BR>==============================================<BR>Initiator
(192.168.1.5) -> Freeswitch(
192.168.1.3):<BR>----------------------------------------------<BR>INVITE
sip:1001@192.168.1.3 SIP/2.0<BR>Via: SIP/2.0/UDP
192.168.1.5:5060;branch=z9hG4bK001834b8b20fdd11b704000fb0e3cf84;rport<BR>From:
"Tosh" <sip:1002@192.168.1.3>;tag=370855464<BR>To:
<sip:1001@192.168.1.3><BR>Call-ID: <A
href="mailto:001834B8-B20F-DD11-B702-000FB0E3CF84@192.168.1.5">001834B8-B20F-DD11-B702-000FB0E3CF84@192.168.1.5</A><BR>CSeq:
98361155 INVITE<BR>Contact:
<sip:1002@192.168.1.5:5060><BR>Proxy-Authorization: (...)<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:
446</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>v=0<BR>o=- 1232061542 0 IN IP4
192.168.1.5<BR>s=SIPPER for PhonerLite<BR>c=IN IP4 192.168.1.5<BR>t=0
0<BR>m=audio 5062 RTP/AVP 0 8 2 3 97 110 101<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:101
telephone-event/8000<BR>a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:3dhne7Il7YqlVZAdnLVgdhngndKXXoNZm7v4/wwC<BR>a=encryption:optional<BR>a=fmtp:101
0-15<BR>a=sendrecv<BR>----------------------------------------------------<BR>Freeswitch
-> Receiver (192.168.1.4)</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>INVITE sip:1001@192.168.1.4:5060 SIP/2.0<BR>Via:
SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKeeFDH2FB5j0Dj<BR>Max-Forwards:
69<BR>From: "Extension 1002"
<sip:1002@192.168.1.3>;tag=ND0tXZH5Qe0aD<BR>To:
<sip:1001@192.168.1.4:5060><BR>Call-ID:
fa523794-8be7-122b-2780-39a48cb53b8d<BR>CSeq: 98362890 INVITE<BR>Contact:
<sip:mod_sofia@192.168.1.3:5060><BR>User-Agent:
FreeSWITCH-mod_sofia/1.0.rc1-7946<BR>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO,
PUBLISH<BR>Supported: 100rel, precondition, timer<BR>Min-SE:
120<BR>Content-Type: application/sdp<BR>Content-Disposition:
session<BR>Content-Length: 428<BR>Remote-Party-ID: "Extension 1002"
<sip:1002@192.168.1.3>;screen=yes;privacy=off</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>v=0<BR>o=FreeSWITCH 5985117983522540515
5861368874018127564 IN IP4 192.168.1.3<BR>s=FreeSWITCH<BR>c=IN IP4
192.168.1.3<BR>t=0 0<BR>a=sendrecv<BR>m=audio 26382 RTP/SAVP 0 9 8 3 101
13<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:9 G722/8000<BR>a=rtpmap:8
PCMA/8000<BR>a=rtpmap:3 GSM/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16<BR>a=rtpmap:13
CN/8000<BR>a=ptime:20<BR>a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:C/XV148O1ZQ0V3LEpByfrFCRL7PGtFDJLcjTCwwV</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial
size=2>------------------------------------------------<BR>Receiver ->
Freeswitch</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
192.168.1.3;rport=5060;branch=z9hG4bKeeFDH2FB5j0Dj<BR>From: "Extension 1002"
<sip:1002@192.168.1.3>;tag=ND0tXZH5Qe0aD<BR>To:
<sip:1001@192.168.1.4:5060>;tag=00c93cd1b20fdd11886f00b0d0b8ce20<BR>Call-ID:
fa523794-8be7-122b-2780-39a48cb53b8d<BR>CSeq: 98362890 INVITE<BR>Contact:
<sip:1001@192.168.1.4:5060><BR>Content-Type: application/sdp<BR>Allow:
INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE<BR>Supported:
replaces, timer<BR>User-Agent: SIPPER for
PhonerLite<BR>Content-Length: 258</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>v=0<BR>o=- 3139884392 1 IN IP4
192.168.1.4<BR>s=SIPPER for PhonerLite<BR>c=IN IP4 192.168.1.4<BR>t=0
0<BR>m=audio 5062 RTP/SAVP 0 8 3 101<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:8
PCMA/8000<BR>a=rtpmap:3 GSM/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101
0-15<BR>a=sendrecv<BR>------------------------------------------------<BR>Freeswitch
-> Initiator</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
192.168.1.5:5060;branch=z9hG4bK001834b8b20fdd11b704000fb0e3cf84;rport=5060<BR>From:
"Tosh" <sip:1002@192.168.1.3>;tag=370855464<BR>To:
<sip:1001@192.168.1.3>;tag=m461U401t59QH<BR>Call-ID: <A
href="mailto:001834B8-B20F-DD11-B702-000FB0E3CF84@192.168.1.5">001834B8-B20F-DD11-B702-000FB0E3CF84@192.168.1.5</A><BR>CSeq:
98361155 INVITE<BR>Contact:
<sip:mod_sofia@192.168.1.3:5060;transport=udp><BR>User-Agent:
FreeSWITCH-mod_sofia/1.0.rc1-7946<BR>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO,
PUBLISH<BR>Supported: 100rel, precondition, timer<BR>Min-SE:
120<BR>Content-Type: application/sdp<BR>Content-Disposition:
session<BR>Content-Length: 155</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>v=0<BR>o=FreeSWITCH 5425860535457980718
3341838566411422164 IN IP4 192.168.1.3<BR>s=FreeSWITCH<BR>c=IN IP4
192.168.1.3<BR>t=0 0<BR>a=sendrecv<BR>m</FONT><FONT face=Arial size=2>=audio 0
RTP/AVP 19</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>=================================================</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>And voice traffic looks like this:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Reciever ->
Freeswitch SRTP</FONT></DIV>
<DIV><FONT face=Arial size=2>Freeswitch ->
Initiator RTP</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I hope this will explain everything. I have also a
wireshark pcap file from this call (but i don't know where and how to send
it). </FONT></DIV>
<DIV><FONT face=Arial size=2>Thanks for help</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=mike@jerris.com href="mailto:mike@jerris.com">Michael Jerris</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> Wednesday, April 23, 2008 9:11
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] SRTP in
PhonerLite and Freeswitch</DIV>
<DIV><BR></DIV>Can you post a sip trace of this entire call, the 19 means we
are rejecting that m= line, are there 2 m lines, AVP and SAVP to indicate
optional secure?
<DIV><BR></DIV>
<DIV>Mike</DIV>
<DIV><BR>
<DIV>On Apr 23, 2008, at 3:01 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 bgcolor="#ffffff">
<DIV><FONT face=Arial size=2>Hi<SPAN
class=Apple-converted-space> </SPAN><BR>I have 2 softphones PhonerLite
(they support SRTP via SDES ) and the freeswitch (windows RC1 version)
server and I wanted to make secure call between those two endpoints
(SRTP).<BR>I spend whole day on testing this scenario and my conclusions
are:<BR>- when the option: <action application="export"
data="sip_secure_media=true"/> is uncommented, and both enpoints have
enabled SRTP then:<BR>1) Initiator of the session sends SIP Invite with
a=crypto paramter and supported codecs<BR>2) Freeswitch receives SIP Invite
and sends SIP Invite to the receiver (also with the crypto)<BR>3) Receiver
receives the SIP Invite with the a=crypto parameter and he sends back
supported codecs with 200 OK message (but without a=crypto parametr. Is that
ok? I'm afraid not)<BR>4) Freeswitch sends 200 OK message but witout any
codecs: m=audio 0 RTP/AVP 19 and no a= parameters!<BR>5) Final result is
that the second leg of the session between Freeswitch and receiver has SRTP
transport enbaled and the first leg (initiator- Freeswitch) doesn't hear
anything - no codecs! However Freeswitch is sending RTP (not SRTP) pacekets
to the initiator.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Could someone explain to me, what is going on,
and why freeswitch doesn't forward codecs accepted by the receiver to the
initiator?<BR>Is it a PhonerLite's bug or freeswitch's? Maybe someone has
tested SRTP with the PhonerLite softphone or any other free softphone with
srtp support?</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>When I uncommented: <param
name="Inbound-no-media" value="true"><BR>everything works fine. The
parameter <action application="export" data="sip_secure_media=true"/>
doesn't change anything then (but i cound miss something).</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Thanks for
help<BR>Chris</FONT></DIV>_______________________________________________<BR>Freeswitch-users
mailing list<BR><A
href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</A><BR><A
href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</A><BR>UNSUBSCRIBE:<A
href="http://lists.freeswitch.org/mailman/options/freeswitch-users">http://lists.freeswitch.org/mailman/options/freeswitch-users</A><BR><A
href="http://www.freeswitch.org">http://www.freeswitch.org</A><BR></DIV></SPAN></BLOCKQUOTE></DIV><BR></DIV>
<P>
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<P></P>_______________________________________________<BR>Freeswitch-users
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