At first glance, it might look like sipX is sending out INVITE's as TLS
but that is not the case. The INVITE is sent out via TCP. The phone
registered its port as 5061 but did not use TLS as the transport so sipX
sends to that port without using TLS. You need to take a look at the
top via and contact as well as "SipUserAgent::sendTcp TCP SIP User
Agent sent message:" which transport is used.
"2011-10-12T14:47:27.468230Z":2912:OUTGOING:INFO:sip1.iptel2.mydomain.com:SipRouter-15:B676AB90:SipXProxy:"SipUserAgent::sendTcp
TCP SIP User Agent sent message:
----Local Host:172.28.246.33---- Port: -1----
----Remote Host:172.28.246.36---- Port: 5061----
INVITE sip:[email protected]:5061;x-sipX-nonat SIP/2.0\r
Record-Route:
<sip:172.28.246.33:5060;lr;sipXecs-CallDest=INT;sipXecs-rs=%2Aauth%7E.%2Afrom%7ENzk4MzdENzItRTVGQjA3%2136350e4507596f8c76ba99490f65b6e4>\r
Via: SIP/2.0/TCP
172.28.246.33;branch=z9hG4bK-XX-0047F3ijOyI0uX0gls58pswfGA\r
Via: SIP/2.0/UDP
172.28.246.33;branch=z9hG4bK-XX-0044JgBDtvG1CD8bDc6`X3TbKQ~Y3eIF7jF_rJYR4dO0jZ5GA\r
Via: SIP/2.0/UDP 172.28.246.35:5061;branch=z9hG4bK1391de18231D1AA\r
From: \"Thure Thuresson\"
<sip:[email protected]>;tag=79837D72-E5FB07\r
To: <sip:[email protected];user=phone>\r
Cseq: 2 INVITE\r
Call-Id: [email protected]\r
Contact: <sip:[email protected]:5061;x-sipX-nonat>\r
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER\r
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.5.0508\r
Accept-Language: en\r
Supported: 100rel,replaces\r
Allow-Events: talk,hold,conference\r
Proxy-Authorization: Digest username=\"203/0004f215c84d\",
realm=\"iptel2.mydomain.com\",
nonce=\"3d233564faa4cee056638d69749e6de04e95a87f\", qop=auth,
cnonce=\"KerlJ7wnVY92ye1\", nc=00000001,
uri=\"sip:[email protected];user=phone\",
response=\"df4878bf32227ef467b50099f838adf0\", algorithm=MD5\r
Max-Forwards: 18\r
Content-Type: application/sdp\r
Content-Length: 441\r
Date: Wed, 12 Oct 2011 14:47:27 GMT\r
Expires: 20\r
\r
v=0\r
o=- 1318430837 1318430837 IN IP4 172.28.246.35\r
s=Polycom IP Phone\r
c=IN IP4 172.28.246.35\r
t=0 0\r
a=sendrecv\r
m=audio 2236 RTP/SAVP 8 0 18 101\r
a=crypto:7 AES_CM_128_HMAC_SHA1_80
inline:wFuzUSUXg0QHPB9ZJ73maU9xc+Z+CXZ7jW+bEC6p\r
a=crypto:8 AES_CM_128_HMAC_SHA1_32
inline:Lek69A/XYS0ooW4hxVJY/XhXOlLuvyazL3TyFKPh\r
a=rtpmap:8 PCMA/8000\r
a=rtpmap:0 PCMU/8000\r
a=rtpmap:18 G729/8000\r
a=fmtp:18 annexb=no\r
a=rtpmap:101 telephone-event/8000\r
--------------------END--------------------"
On 10/13/2011 01:57 AM, Sebastian wrote:
Hello again!
Thank you very much for your fast answers!
I attach the log file, and also uploads two captures from
wireshark.
As I said, only the traffic between SipX and destination
phone is encrypted, no matter who the destination phone is.
If I make a call from the other phone (prev. dest. phone),
which just recently had encrypted communication with SipX,
you can see that the traffic between the two now goes
unencrypted, while the traffic between SipX and the "new"
dest. phone is encrypted. The pictures explains it. I think
this i very strange, but maybe I'm reading it wrong?
(My SipX-server is .33)
Making a call from 202(.36) to 203(.35):
Making a call from 203(.35) to 202(.36):
When I, in the SipXecs web-GUI or on the phone web-GUI,
configure the phones to use TLS and port 5061 to talk to
their registrar, they simply cannot register. So at the
moment they use UDP.
Thanks.
//SebbJ
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