Chris,
Heiko and I have had a few exchanges in email... He's a very nice guy
very preceptive to the changes required to support this properly. He
said he'll be trying out the two m-line method soon.
/b
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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
172.16.115.114
:5070;branch=z9hG4bK0014de144110dd118b3d00123fa91863;rport
From: <sip:[EMAIL PROTECTED]>;tag=3538012975
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Contact: <sip:[EMAIL PROTECTED]: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: [email protected]
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:[EMAIL PROTECTED]
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Brian West
sip:[EMAIL PROTECTED]
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