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@172.16.115.114CSeq:
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 "
----- Original Message -----
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.
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