Hello,
I’m experimenting with Asterisk and TLS/SRTP.
When connecting a Nokia e72 phone everything is fine as long as I’m not using security (SIP over UDP and RTP).
When I enable TLS the nokia starts sending invalid INVITE request:
<--- SIP read from TLS:192.168.100.1:58385 --->
INVITE sips:home.ilian.org:5061 SIP/2.0
Route: <sips:600@home.ilian.org>
Via: SIP/2.0/TLS 192.168.100.100:5061;branch=z9hG4bK2ba9ok5thhhc6luc062goh4;rport
From: <sips:e72@home.ilian.org>;tag=podpok312thc6qcp062g
To: <sips:600@home.ilian.org>
Contact: <sips:cs2EIXlmpxk-HNysbdrd@192.168.100.100:5061>
Supported: precondition,100rel,timer,sec-agree
CSeq: 4771 INVITE
Call-ID: 6Fqsk0DWoIfuaw653bJt-waFHyAhQe
Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
User-Agent: Nokia RM-530 052.005 (en)
Expires: 120
Privacy: None
Session-Expires: 1800
Max-Forwards: 70
Content-Type: application/sdp
Accept-Language: en
Content-Length: 803
v=0
o=e72 63458720826586125 63458720826586125 IN IP4 192.168.100.100
s=-
c=IN IP4 192.168.100.100
t=0 0
m=audio 49152 RTP/SAVP 100 96 0 8 97 18 98 13
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:RGMtdWtVN0ZZS2lFVDBJTzBCSHpwSVpNcWNXbmtO
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:c25jOEdVbm0tU09UTDdLbnV1ZnVTanBia1oyVUk3
a=curr:sec e2e none
a=des:sec optional e2e sendrecv
a=rtcp:49153 IN IP4 192.168.100.100
a=rtpmap:100 AMR-WB/16000
a=ptime:20
a=maxptime:200
a=fmtp:100 mode-change-period=2; mode-change-neighbor=1
a=rtpmap:96 AMR/8000
a=fmtp:96 mode-set=0,1,2,3,4,5,6,7; mode-change-neighbor=1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-15
a=rtpmap:13 CN/8000
<------------->
--- (18 headers 26 lines) ---
Sending to 192.168.100.1:58385 (NAT)
Using INVITE request as basis request - 6Fqsk0DWoIfuaw653bJt-waFHyAhQe
Found peer 'e72' for 'e72' from 192.168.100.1:58385
== Using SIP RTP CoS mark 5
Found RTP audio format 100
Found RTP audio format 96
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 98
Found RTP audio format 13
Found audio description format AMR-WB for ID 100
Found audio description format AMR for ID 96
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 97
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 98
Found audio description format CN for ID 13
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.100.100:49152
Looking for in default (domain home.ilian.org:5061)
<--- Reliably Transmitting (NAT) to 192.168.100.1:58385 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TLS 192.168.100.100:5061;branch=z9hG4bK2ba9ok5thhhc6luc062goh4;received=192.168.100.1;rport=58385
From: <sips:e72@home.ilian.org>;tag=podpok312thc6qcp062g
To: <sips:600@home.ilian.org>;tag=as496f10d5
Call-ID: 6Fqsk0DWoIfuaw653bJt-waFHyAhQe
CSeq: 4771 INVITE
Server: Asterisk PBX SVN-trunk-r295748
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
This is parsed as the nokia trying to call: home.ilian.org instead of extention 600. Can anyone give me an idea how can I fix this?